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

[alsa-devel] [PATCH 1/9] R3Di and SBZ quirk entires + alt firmware loading
by Connor McAdams 04 May '18

04 May '18

This patch adds PCI quirk ID's for the Sound Blaster Z and Recon3Di. Only the currently tested ID's have been added. This patch also adds the ability to load alternative firmwares for each card, the firmwares can be obtained from within the Windows driver. The Recon3Di uses "ctefx-r3di.bin" and the Sound Blaster Z uses "ctefx-sbz.bin". If the alternative firmware for the given quirk is not found, the original ctefx.bin will be used. This has been confirmed to work for both the R3Di and the SBZ. This patch also makes the character array *dirstr a const. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 61 +++++++++++++++++++++++++++++++++++++++----- 1 file changed, 55 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea86..8346100 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -72,12 +72,16 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *dirstr[2] = { "Playback", "Capture" }; enum { SPEAKER_OUT, @@ -734,6 +738,7 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; /* mixer and effects related */ unsigned char dmic_ctl; @@ -762,6 +767,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -782,6 +789,10 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -3207,7 +3218,7 @@ static int ca0132_select_out(struct hda_codec *codec) pin_ctl & ~PIN_HP); /* enable speaker node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); snd_hda_set_pin_ctl(codec, spec->out_pins[0], pin_ctl | PIN_OUT); } else { @@ -4370,11 +4381,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* + * Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (!spec->alt_firmware_present) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { @@ -4476,7 +4525,7 @@ static struct hda_verb ca0132_base_exit_verbs[] = { {} }; -/* Other verbs tables. Sends after DSP download. */ +/* Other verbs tables. Sends after DSP download. */ static struct hda_verb ca0132_init_verbs0[] = { /* chip init verbs */ {0x15, 0x70D, 0xF0}, -- 2.7.4

3 11

[alsa-devel] [PATCH 1/9] R3Di and SBZ quirk entires + alt firmware loading
by Connor McAdams 04 May '18

04 May '18

This patch adds PCI quirk ID's for the Sound Blaster Z and Recon3Di. Only the currently tested ID's have been added. This patch also adds the ability to load alternative firmwares for each card, the firmwares can be obtained from within the Windows driver. The Recon3Di uses "ctefx-r3di.bin" and the Sound Blaster Z uses "ctefx-sbz.bin". If the alternative firmware for the given quirk is not found, the original ctefx.bin will be used. This has been confirmed to work for both the R3Di and the SBZ. This patch also makes the character array *dirstr a const. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 55 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea86..9a0286e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -72,12 +72,16 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *dirstr[2] = { "Playback", "Capture" }; enum { SPEAKER_OUT, @@ -734,6 +738,7 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; /* mixer and effects related */ unsigned char dmic_ctl; @@ -762,6 +767,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -782,6 +789,10 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -4370,11 +4381,47 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (spec->alt_firmware_present == false) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { -- 2.7.4

2 9

[alsa-devel] Segmentation fault in libasound.so.2.0.0 (alsa-lib-1.1.6)
by Paul Bordukoff 04 May '18

04 May '18

Hello everyone, I am using ALSA (alsa-lib-1.1.6) on Gentoo laptop with x32 ABI (linux-4.16.7). My mpv (0.27.2) gets segfault each time I try to play video with sound: pashgan@light ~/temp $ mpv ./SampleVideo_1280x720_10mb.mp4 Playing: ./SampleVideo_1280x720_10mb.mp4 (+) Video --vid=1 (*) (h264 1280x720 25.000fps) (+) Audio --aid=1 --alang=und (*) (aac 6ch 48000Hz) AO: [alsa] 48000Hz stereo 2ch float VO: [opengl] 1280x720 yuv420p Segmentation fault dmesg shows: [12526.319909] traps: mpv/ao[13621] general protection ip:f28ce4fa sp:e46f2d60 error:0 in libasound.so.2.0.0[f2843000+fe000] gdb session: (gdb) bt #0 0xf29b14fa in mix_areas_32_smp (size=3935624896, dst=0x56a6e7f0, src=0x56a6e7f0, sum=0x1, dst_step=4, src_step=4, sum_step=4) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix_x86_64.h:143 #1 0xf29b2361 in mix_areas (size=3935493824, dst_ofs=4056, src_ofs=0, dst_areas=0x56a6cd90, src_areas=0x56a6dc30, dmix=0x56a6c900) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:197 #2 snd_pcm_dmix_sync_area (pcm=pcm@entry=0x56a6d270) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:379 #3 0xf29b267f in snd_pcm_dmix_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_dmix.c:611 #4 0xf2972349 in __snd_pcm_start (pcm=0x56a6d270, pcm=0x56a6d270) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #5 snd_pcm_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:1226 #6 0xf2972349 in __snd_pcm_start (pcm=0x56a6d8b0, pcm=0x56a6d8b0) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #7 snd_pcm_start (pcm=<optimized out>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:1226 #8 0xf2979a78 in __snd_pcm_start (pcm=0x56a52e60, pcm=0x56a52e60) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_local.h:434 #9 snd1_pcm_write_areas (pcm=pcm@entry=0x56a52e60, areas=areas@entry=0xe47f2f00, offset=offset@entry=0, size=<optimized out>, size@entry=12288, func=func@entry=0xf2988770 <snd_pcm_plugin_write_areas>) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm.c:7421 #10 0xf29890e7 in snd_pcm_plugin_writei (pcm=<optimized out>, buffer=<optimized out>, size=12288) at /tmp/portage/media-libs/alsa-lib-1.1.6-r1/work/alsa-lib-1.1.6/src/pcm/pcm_plugin.c:358 #11 0x56590280 in ?? () #12 0x56594846 in ?? () #13 0xf1b9808b in start_thread () from /libx32/libpthread.so.0 #14 0xf18d4e2f in clone () from /libx32/libc.so.6 (gdb) Same bug in Gentoo bug tracker: https://bugs.gentoo.org/654754 I need help to investigate pcm_dmix behaviour, I see asm code there (pcm_dmix_x86_64.h:143) but have no idea how can I debug it (my asm experience is far from ok). Can somebody guide me?

1 0

[alsa-devel] [PATCH v4] ASoC: tas6424: Allow disabling auto diagnostics for faster power-on
by Jean-Jacques Hiblot 04 May '18

04 May '18

The TAS6424 incorporates both DC-load and AC-load diagnostics which are used to determine the status of the load. The DC diagnostics runs when any channel is directed to leave the Hi-Z state and enter the MUTE or PLAY state. The DC diagnostics are turned on by default but, if a fast startup without diagnostics is required, the diagnostics can be disabled using a dedicated ALSA control. Signed-off-by: Jean-Jacques Hiblot <jjhiblot(a)ti.com> --- changes in v4: - rename the control "Auto Diagnostics Swicth" with a capital S on the Switch sound/soc/codecs/tas6424.c | 13 +++++++++++-- sound/soc/codecs/tas6424.h | 4 ++++ 2 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 89fd0c1..14999b9 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -64,6 +64,8 @@ static const struct snd_kcontrol_new tas6424_snd_controls[] = { TAS6424_CH3_VOL_CTRL, 0, 0xff, 0, dac_tlv), SOC_SINGLE_TLV("Speaker Driver CH4 Playback Volume", TAS6424_CH4_VOL_CTRL, 0, 0xff, 0, dac_tlv), + SOC_SINGLE_STROBE("Auto Diagnostics Switch", TAS6424_DC_DIAG_CTRL1, + TAS6424_LDGBYPASS_SHIFT, 1), }; static int tas6424_dac_event(struct snd_soc_dapm_widget *w, @@ -297,6 +299,11 @@ static int tas6424_power_on(struct snd_soc_component *component) struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); int ret; u8 chan_states; + int no_auto_diags = 0; + unsigned int reg_val; + + if (!regmap_read(tas6424->regmap, TAS6424_DC_DIAG_CTRL1, &reg_val)) + no_auto_diags = reg_val & TAS6424_LDGBYPASS_MASK; ret = regulator_bulk_enable(ARRAY_SIZE(tas6424->supplies), tas6424->supplies); @@ -327,9 +334,11 @@ static int tas6424_power_on(struct snd_soc_component *component) snd_soc_component_write(component, TAS6424_CH_STATE_CTRL, chan_states); /* any time we come out of HIZ, the output channels automatically run DC - * load diagnostics, wait here until this completes + * load diagnostics if autodiagnotics are enabled. wait here until this + * completes. */ - msleep(230); + if (!no_auto_diags) + msleep(230); return 0; } diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index 4305883..b5958c4 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -111,6 +111,10 @@ TAS6424_CH3_STATE_DIAG | \ TAS6424_CH4_STATE_DIAG) +/* TAS6424_DC_DIAG_CTRL1 */ +#define TAS6424_LDGBYPASS_SHIFT 0 +#define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT) + /* TAS6424_GLOB_FAULT1_REG */ #define TAS6424_FAULT_CLOCK BIT(4) #define TAS6424_FAULT_PVDD_OV BIT(3) -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Clean up mixer element list traverse
by Takashi Iwai 03 May '18

03 May '18

Introduce a new macro for iterating over mixer element list for avoiding the open codes in many places. Also the open-coded container_of() and the forced cast to struct usb_mixer_elem_info are replaced with another simple macro, too. No functional changes but just readability improvement. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 20 +++++++++----------- sound/usb/mixer.h | 6 ++++++ sound/usb/mixer_quirks.c | 2 +- sound/usb/mixer_scarlett.c | 6 ++---- 4 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index bb203b3684fc..265258b0e74c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2598,9 +2598,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = - (struct usb_mixer_elem_info *)list; + mixer_elem_list_to_info(list); /* invalidate cache, so the value is read from the device */ info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, @@ -2611,7 +2611,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " @@ -2637,8 +2637,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (list = mixer->id_elems[unitid]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { snd_iprintf(buffer, " Unit: %i\n", list->id); if (list->kctl) snd_iprintf(buffer, @@ -2668,19 +2667,19 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) count++; if (count == 0) return; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info; if (!list->kctl) continue; - info = (struct usb_mixer_elem_info *)list; + info = mixer_elem_list_to_info(list); if (count > 1 && info->control != control) continue; @@ -2946,7 +2945,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) static int restore_mixer_value(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; if (cval->cmask) { @@ -2982,8 +2981,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (list = mixer->id_elems[id]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, id) { if (list->resume) { err = list->resume(list); if (err < 0) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ba27f7ade670..e02653465e29 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -53,6 +53,12 @@ struct usb_mixer_elem_list { usb_mixer_elem_resume_func_t resume; }; +/* iterate over mixer element list of the given unit id */ +#define for_each_mixer_elem(list, mixer, id) \ + for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem) +#define mixer_elem_list_to_info(list) \ + container_of(list, struct usb_mixer_elem_info, head) + struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4377374affd3..1b94387e18b6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1172,7 +1172,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index c33e2378089d..4aeb9488a0c9 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); int i; for (i = 0; i < elem->channels; i++) @@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); if (elem->cached) snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); -- 2.16.3

1 0

[alsa-devel] Rationale for BYTCR defaults in the kernel's bytcr_rt5640 machine driver ?
by Hans de Goede 03 May '18

03 May '18

Hi Pierre-Louis, I'm finally wrapping up my rt5640 jack-detect work, as such I'm currently testing speaker + headphones + internal mic + headset-mic + jack-detect functionality on the 10 different x86 devices with a rt5640 codec which I've gathered. 2 of them stand out in that they use the BYTCR SoC, but don't have the ACPI table for detecting if SSP0 AIF1 or AIF2 should be used. Currently the driver defaults to AIF2 in this case. As the somewhat wildcard-ish DMI quirk for boards where the sys_vendor is "Insyde" (which applies to a lot of generic designs) shows: { .callback = byt_rt5640_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | BYT_RT5640_MCLK_EN | BYT_RT5640_SSP0_AIF1), }, AIF1 seems to be a better default both models without the ACPI table which I've here, a HP pavilion X2 and a Toshiba Click Mini L9W-B need a quirk to use AIF1. So I was wondering if there was a specific rationale for the AIF2 default and if it would not be better to change the default to AIF1 (which will unfortunately bring a chance of regressions) ? Regards, Hans

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[alsa-devel] [BUG - REPORT] ALC892 not working
by Xristos Boul 03 May '18

03 May '18

Hi everyone, I'm having a problem with my line-out port not being working properly. Specifically it can't play audio when the 'alsamixer' auto-mute mode is enabled, so sound is coming only from the headphones.However, when auto-mute is disabled the sound is coming from both ports simultaneously. I'm using the Asus Z170-A mobo with ALC892 codec, OS: Arch I ran alsa-info.sh on both cases and i saw that the Pin-Complex 0x14 had 'Pin-ctls: OUT' when auto-mute was disabled whereas when auto-mute was enabled it was 'Pin-ctls: 0x00' (as it currenly is). So i ran HDA-analyzer tool and changed that widget setting to 'OUT' and obviously i produced the same behaviour like when i disabled 'auto-mute' (again sound coming from both ports) and also in all cases PulseAudio lists the 'Line-out' port as 'unplugged'. Also 2 more things: 1st: In BIOS when i changed the driver for the Front panel from 'HD-Audio' to 'AC97' and the ports worked correctly, however no 'auto-mute' mode was available so i had to mute the ports manually when i plugged/unplugged the headphones for e.x I tried uninstalling PulseAudio to see if it was the case but the problem persisted so i think it really is some codec problem since i also have a similar problem in Windows OS. 2nd: When i unplugged the Front-panel connection from the mobo the port played correctly. So is it maybe a mixer problem or something related? Finally i tried every 'model' parameter from HD-Audio-Models.txt for ALC892 but got nothing. In any case i'm attaching the current alsa-info.sh. If you would like i could also send alsa-info.sh for all those cases above. I don't think i'm alone in this issue but i would greatly appreciate the help! Thanks in advance. Link to alsa-info.txt -> http://www.alsa-project.org/db/?f=1020241a74addd837a3625d3f1c6be374996aa71

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[alsa-devel] [PATCH] ALSA: sparc: Use GFP_KERNEL for non-atomic allocation
by Takashi Iwai 03 May '18

03 May '18

dbri driver allocates a resource with GFP_ATOMIC unnecessarily in its probe function. Replace it with the standard GFP_KERNEL for avoiding the bogus allocation failures. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/sparc/dbri.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index abc7bd5055eb..f0e713527e91 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2542,7 +2542,7 @@ static int snd_dbri_create(struct snd_card *card, dbri->irq = irq; dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_ATOMIC); + &dbri->dma_dvma, GFP_KERNEL); if (!dbri->dma) return -ENOMEM; -- 2.16.3

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[alsa-devel] [PATCH v2] ASoC: da7219: read fmw property to get mclk for non-dts systems
by Akshu Agrawal 03 May '18

03 May '18

Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Fixed kbuild error include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16..df7ddf4 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8..aed68a4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1624,6 +1624,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone dev_warn(dev, "Using default clk name: %s\n", pdata->dai_clks_name); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1905,7 +1907,10 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = devm_clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 1.9.1

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[alsa-devel] [PATCH alsa-lib] conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio
by Takashi Iwai 03 May '18

03 May '18

USB-audio device on Dell WD15 docking station provides two individual PCM streams, one for headphone and another for line out. A UCM profile gives the proper roles for these. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is the alsa-lib part to correspond to the recent Dell WD15 dock patchset. src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf | 5 +++++ src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 26 +++++++++++++++++++++++++ src/conf/ucm/Dell-WD15-Dock/Makefile.am | 4 ++++ src/conf/ucm/Makefile.am | 1 + 4 files changed, 36 insertions(+) create mode 100644 src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf create mode 100644 src/conf/ucm/Dell-WD15-Dock/HiFi.conf create mode 100644 src/conf/ucm/Dell-WD15-Dock/Makefile.am diff --git a/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf new file mode 100644 index 000000000000..f4f0a8d97e73 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf @@ -0,0 +1,5 @@ +Comment "USB-audio on Dell WD15 docking station" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf new file mode 100644 index 000000000000..b07f34400086 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -0,0 +1,26 @@ +SectionDevice."Headphone" { + Comment "Headphone" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:WD15Dock,0" + } +} + +SectionDevice."LineOut" { + Comment "Line Out" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:WD15Dock,1" + } +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CaptureChannels "2" + CapturePCM "hw:WD15Dock,0" + } +} diff --git a/src/conf/ucm/Dell-WD15-Dock/Makefile.am b/src/conf/ucm/Dell-WD15-Dock/Makefile.am new file mode 100644 index 000000000000..7ab58730dc26 --- /dev/null +++ b/src/conf/ucm/Dell-WD15-Dock/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/Dell-WD15-Dock +ucm_DATA = Dell-WD15-Dock.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 02df25f01b25..38fc6b252bcb 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -7,6 +7,7 @@ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ DAISY-I2S \ DB410c \ +Dell-WD15-Dock \ GoogleNyan \ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ -- 2.16.3

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