September 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rockchip: add missing INTERLEAVED PCM attribute
by Katsuhiro Suzuki 24 Apr '19

24 Apr '19

This patch adds SNDRV_PCM_INFO_INTERLEAVED into PCM hardware info. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index f77538319221..9e7b5fa4cf59 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, -- 2.18.0

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[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

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[alsa-devel] snd_hda_codec_hdmi: `hdaudio hdaudioC0D2: Unable to bind the codec`
by Paul Menzel 18 Feb '19

18 Feb '19

Dear Linux folks, With drm-tip (v4.19-rc3-946-g09b295662edd) Linux prints `snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec` to the log. ``` […] [ 12.481788] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC671: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 12.482536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) [ 12.483184] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 12.483737] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.484198] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.484582] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 12.485100] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 12.485519] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 12.497685] snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec [ 12.498311] hdaudio hdaudioC0D2: Unable to bind the codec [ 12.498789] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input6 [ 12.499567] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input7 [ 12.500268] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input8 [ 12.501039] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 12.501683] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 17.743338] [drm:intel_pch_type [i915]] Found SunrisePoint PCH […] ``` Is that an error or just a notice? Kind regards, Paul

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

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[alsa-devel] [PATCH 0/6] ASoC: rt5651: Various fixes and quirks
by Hans de Goede 24 Dec '18

24 Dec '18

Hi All, Here is my last (big) round of (Intel) rt5651 support updates, with these patches combined with the updated, matching UCM profiles from here: https://github.com/jwrdegoede/alsa-lib/commits/master All 7 different devices I have work 100% OOTB, without requiring any manual config or mixer setup. After this there should be no more big changes. I will submit the matching alsa-lib UCM profile changes upstream once this set is in -next (and thus I can be sure the long-names will not change due to review). Note that unfortunately my previous round of cleanup patches removed 1 input mapping (of the 3 mappings in total it removed) which is actually necessary. So this series re-introduces that 1 mapping. I did not notice this before because UCM input profile switching does not work reliably for some reason, something which I still need to investigate. This time I manually veryfied that the correct / expected mixer settings where used while testing. Regards, Hans

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[alsa-devel] [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by Hans de Goede 28 Nov '18

28 Nov '18

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is identical to all the cset statements this commit removes. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/chtrt5645/HiFi.conf | 116 +------------------------------ 1 file changed, 1 insertion(+), 115 deletions(-) diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index e81866cf..d993f6ae 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,121 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - # media mixer settings - # compress - cset "name='media0_in Gain 0 Switch' on" - cset "name='media0_in Gain 0 Volume' 0" - - # normal - cset "name='media1_in Gain 0 Switch' on" - cset "name='media1_in Gain 0 Volume' 0" - # swm loopback - cset "name='media2_in Gain 0 Switch' off" - cset "name='media2_in Gain 0 Volume' 0%" - # deep buffer - cset "name='media3_in Gain 0 Switch' on" - cset "name='media3_in Gain 0 Volume' 0" - - cset "name='media0_out mix 0 media0_in Switch' on" - cset "name='media0_out mix 0 media1_in Switch' on" - cset "name='media0_out mix 0 media2_in Switch' off" - cset "name='media0_out mix 0 media3_in Switch' on" - - cset "name='media1_out mix 0 media0_in Switch' off" - cset "name='media1_out mix 0 media1_in Switch' off" - cset "name='media1_out mix 0 media2_in Switch' off" - cset "name='media1_out mix 0 media3_in Switch' off" - - cset "name='pcm0_in Gain 0 Switch' on" - cset "name='pcm0_in Gain 0 Volume' 0" - - cset "name='pcm1_in Gain 0 Switch' off" - cset "name='pcm1_in Gain 0 Volume' 0%" - - # codec0_out settings (used if ssp2 is connected to aif1) - cset "name='codec_out0 mix 0 codec_in0 Switch' off" - cset "name='codec_out0 mix 0 codec_in1 Switch' off" - cset "name='codec_out0 mix 0 media_loop1_in Switch' off" - cset "name='codec_out0 mix 0 media_loop2_in Switch' off" - cset "name='codec_out0 mix 0 pcm0_in Switch' on" - cset "name='codec_out0 mix 0 pcm1_in Switch' off" - cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out0 Gain 0 Switch' on" - cset "name='codec_out0 Gain 0 Volume' 0" - - # modem_out settings (used if ssp0 is connected to aif2) - cset "name='modem_out mix 0 codec_in0 Switch' off" - cset "name='modem_out mix 0 codec_in1 Switch' off" - cset "name='modem_out mix 0 media_loop1_in Switch' off" - cset "name='modem_out mix 0 media_loop2_in Switch' off" - cset "name='modem_out mix 0 pcm0_in Switch' on" - cset "name='modem_out mix 0 pcm1_in Switch' off" - cset "name='modem_out mix 0 sprot_loop_in Switch' off" - cset "name='modem_out Gain 0 Switch' on" - cset "name='modem_out Gain 0 Volume' 0" - - # input settings - # pcm1_out settings - - # input used when SSP2 is connected - cset "name='codec_in0 Gain 0 Switch' on" - cset "name='codec_in0 Gain 0 Volume' 0" - - # input used when SSP0 is connected - cset "name='modem_in Gain 0 Switch' on" - cset "name='modem_in Gain 0 Volume' 0" - - cset "name='pcm1_out mix 0 codec_in0 Switch' on" - cset "name='pcm1_out mix 0 modem_in Switch' on" - cset "name='pcm1_out mix 0 codec_in1 Switch' off" - cset "name='pcm1_out mix 0 media_loop1_in Switch' off" - cset "name='pcm1_out mix 0 media_loop2_in Switch' off" - cset "name='pcm1_out mix 0 pcm0_in Switch' off" - cset "name='pcm1_out mix 0 pcm1_in Switch' off" - cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" - - cset "name='pcm1_out Gain 0 Switch' on" - cset "name='pcm1_out Gain 0 Volume' 0" - - # disable codec_out1 - cset "name='codec_out1 mix 0 codec_in0 Switch' off" - cset "name='codec_out1 mix 0 codec_in1 Switch' off" - cset "name='codec_out1 mix 0 media_loop1_in Switch' off" - cset "name='codec_out1 mix 0 media_loop2_in Switch' off" - cset "name='codec_out1 mix 0 pcm0_in Switch' off" - cset "name='codec_out1 mix 0 pcm1_in Switch' off" - cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out1 Gain 0 Switch' off" - cset "name='codec_out1 Gain 0 Volume' 0%" - - # disable codec_in1 - cset "name='codec_in1 Gain 0 Switch' off" - cset "name='codec_in1 Gain 0 Volume' 0%" - - # disable all loops - cset "name='media_loop1_out mix 0 codec_in0 Switch' off" - cset "name='media_loop1_out mix 0 codec_in1 Switch' off" - cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop1_out mix 0 pcm0_in Switch' off" - cset "name='media_loop1_out mix 0 pcm1_in Switch' off" - cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" - - cset "name='media_loop2_out mix 0 codec_in0 Switch' off" - cset "name='media_loop2_out mix 0 codec_in1 Switch' off" - cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop2_out mix 0 pcm0_in Switch' off" - cset "name='media_loop2_out mix 0 pcm1_in Switch' off" - cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" - - cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" - cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" - cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" - cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" - cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + <bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" -- 2.18.0

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Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 27 Nov '18

27 Nov '18

Please don't drop Cc to ML. Also avoid top-posting. On Sat, 04 Aug 2018 19:02:53 +0200, David Ulricht wrote: > > How can I do that ? > I tried verifying if /proc/asound/card0/codec#0 changes when i put > handphone jack in and out with "diff ver1 ver2" and it is exactly the same. Just do for each bit as you tried for GPIO output. In this case, the direction bit should be different (set 0), and you need to read. > > One strange thing I note is hda-jack-retask.fw contains: > [pincfg] > 0x24 0x90100080 > 0x25 0x90100082 > I have 0x90100080 for 0x24 but in /proc/asound/card0/codec#0 : > Node 0x24 [Pin Complex] wcaps 0x400101: Stereo > Pincap 0x00000010: OUT > Pin Default 0x90100110: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x02 > > Shouldn't "Pin Default 0x90100110" be 0x90100080 after applying the patch ? No, the codec config value itself won't change, the driver uses the given value only internally. Takashi > I can see that Applying patch firmware 'hda-jack-retask.fw' is before > hdaudioC0D0: autoconfig for Generic: line_outs=2 (0x24/0x25/0x0/0x0/0x0) > type:speaker > in dmesg and this worries me, maybe the pins are overriden by generic > cirrus logic ? > > > On Sat, Aug 4, 2018 at 7:26 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 17:44:59 +0200, > > David Ulricht wrote: > > > > > > I used the following script running as root to verify the 8 GPIOs while > > > playing long youtube video, > > > I have running pulseaudio meanwhile and not restarting it, i think its > > okay > > > like that people on the ubuntu forums say that after using > > > hda-verb and setting a gpio active immediately sound appears, so i guess > > > there is no need to restart pulseaudio. > > > I have done the same test with headphones in. > > > No sound in both scenarios, no input no output detected (i'm watching > > > pavucontrol for input detection). > > > > This codec is fairly unique and doesn't provide the standard jack > > detection on each pin. Maybe it's via either GPIO or any other > > method. You can try to read GPIO pins (set as input) to see whether > > any of them corresponds to the jack detection, for example. > > > > > > Takashi > > > > > > > > #!/usr/bin/env python3 > > > import os,time > > > for x in range(0,256): > > > mhex=("0x%0.2X" % x) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > > > time.sleep(3) > > > > > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > > > IO[0-7] changes its configuration, > > > in comparison Joel Krahemann in his scripts is changing the hex right > > after > > > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > > > correctly. > > > > > > I found the following post: > > > https://forum.manjaro.org/t/sound-not-working-on-a-2017- > > imac-27-5k-retina/43638 > > > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 > > has > > > a similar soundcard. > > > How can i setup with command line or any other utility: > > > """select “off” in the top menu that propose to setup the Digital > > > Surround""" > > > this is some setting in xfce4-mixer package which is not available for > > > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > > > find this setting in audio mixer of xfce4 either. > > > > > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > > > David Ulricht wrote: > > > > > > > > > > Hello, > > > > > > > > > > Macbook pro models containing CS8409 for sure are: > > > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > > > > > I'm attaching working sound configuration from Windows 10 registry > > (note > > > > > that the cs420x might need to be ignored or might be important i > > don't > > > > > really know). I believe what is really used is in the > > PinConfigOverride > > > > > section. > > > > > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > > > and your override (user_pin_cfg), there aren't so many changes. > > > > All changes are pretty minor, and I guess it won't influence on the > > > > driver behavior. > > > > > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > > > Or it's some direct I2C control. The INI file mentions it. > > > > And that's above HD-audio driver's responsibility. > > > > > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > > > > > It's because the codec chip has no output amplifier at all. > > > > So there can be neither output volume nor mute control on this chip. > > > > > > > > > What I have tried is execute: > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > > > putting @@ from 0x00 to 0x50 > > > > > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > > > 0x02, 0x04, 0x08, ... 0x80. > > > > > > > > And how is the current situation? You can't play *and* record > > > > anything from any inputs / outputs? > > > > > > > > > > > > Takashi > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [PATCH 1/2] ASoC: max9892x: Add documentation for reset-gpio support
by Cheng-Yi Chiang 20 Nov '18

20 Nov '18

max98927 codec driver will support reset-gpio binding so it can toggle reset line in its probe function. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/max9892x.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt index f6171591ddc61..8bf30483eedcd 100644 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -29,6 +29,8 @@ Required properties: Range : 0 (off), 1 (on), Default : 0 - reg : the I2C address of the device for I2C + - reset-gpio: reset gpio the chip is connected to. This is supported + on max98927. Example: @@ -38,4 +40,5 @@ codec: max98927@3a { imon-slot-no = <1>; interleave-mode = <0>; reg = <0x3a>; + reset-gpio = <&tlmm 75 GPIO_ACTIVE_LOW>; }; -- 2.16.4

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Re: [alsa-devel] Controlling the Tascam FW-1884
by Takashi Sakamoto 16 Nov '18

16 Nov '18

Hi Scott, (C.Ced to alsa-devel, ffado-devel) On Sep 7 2018 23:01, Scott Bahling wrote: > I recently got my hands on a FW-1884 and would like to get it working in > Linux. I'm mostly interested in using the control interface which should > support several midi control interface protocols like Mackie Control or > Mackie HUI. From the user manual it appears that the host computer needs to > set the registers on the FW-1884 to switch modes (they reference Windows > applications). > > I see that you have recently been working on user space support asynchronous > commands to such devices. The hinawa_utils code has some beginnings of > support for setting options on the Tascam devices and I have just started > playing with that. Thanks for your suggestion. At present, the most of features on control surface of FW-1884 are not available, especially for fader, knob and start/stop buttons. In data transmission between system and devices on IEEE 1394 bus, both of asynchronous transactions and isochronous packet streams are available. In a case of TASCAM FireWire series, messages via MIDI physical interface is bypassed to asynchronous transaction, while messages generated on control surface are bypassed to isochronous packets, multiplexed with PCM frames. When enabling multiplexing messages to isochronous packet, demultiplexer should be implemented to ALSA firewire-tascam driver, but not yet. You can see my initial RFC in alsa-devel[1]. This patch enables userspace applications to perform the demultiplexing, however this idea easily brings cache-coherency problems due to page mapping between kernel/userspace and impossible to be applied. We need to investigate another idea to achieve the demultiplexing in kernel land. As another option, according to page 26 in 'TASCAM FW-1884 Owner's Manual', when programming Keys to port 1-4, system could receive MIDI messages from control surface of FW-1884. However, I've not investigated it. > How are you determining the registers and flags? Do you have documentation, > or are you reverse engineering? Do you know the registers and flags for > setting the control protocol? > > If you have any hints on how figure out the proper registers and flags I'm > happy to contribute to the work you have started and help support more > functions of the FW-1884. User manuals are documents to which we can refer. The most vendors licenses their drivers/software with EULA and in most cases it can't allow users to binary-based reverse engineering. All of my work are done with external protocol analyzer unit on IEEE 1394 bus. When working to implement drivers, I read/classified/parsed content of many packets sniffed by the analyzer. An initial idea of libhinawa/hinawa-utils is to assist this my work. I heard FFADO developers uses some ways for this type of work: - Usage of host controller of TI PCILynx and driver module (nosy.ko) - Windows application; bushound[2] However, this type of work takes you long time, so I don't necessarily recommend you. It's 2018 and I think you can find alternative control surfaces on USB. If you take long time to make devices available on GNU/Linux, it's better to select devices on 'live' buses instead of 'legacy' buses such as IEEE 1394 bus, IMO. In a case of USB, wireshark with usbcap is the easiest way for us to sniff packets on Windows. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://perisoft.net/bushound/ Regards Takashi Sakamoto

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