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

[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

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[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

1 0

[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

1 0

[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

5 9

[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

2 4

[alsa-devel] [PATCH] conf/ucm: Fix build error for UCM Dell WD15 Dock USB-audio
by Xiuli Pan 03 May '18

03 May '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> Miss oneline in configure.ac in provious patch conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- configure.ac | 1 + 1 file changed, 1 insertion(+) diff --git a/configure.ac b/configure.ac index 3ee989e..693b5d5 100644 --- a/configure.ac +++ b/configure.ac @@ -725,6 +725,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/Dell-WD15-Dock/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt565: fix missing assignment to ret_val
by Colin King 03 May '18

03 May '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the check that ret_val is not -ENOENT is always true and the quirk bit BYT_RY5651_MCLK_EN is never being cleared because ret_val is always zero at this point from a previous assignment earlier on. I believe that ret_val should actually be assigned to the return from devm_clk_get() as this can return -ENOENT (from a deeper call to clk_get_sys) and that was the original intention to check this. Detected by CoverityScan, CID#1460228 ("Logically dead code") Fixes: 02c0a3b3047f ("ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3c7d93520c52..469e2695d121 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -856,9 +856,10 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); /* * Fall back to bit clock usage for -ENOENT (clock not * available likely due to missing dependencies), bail -- 2.17.0

3 2

[alsa-devel] [PATCH v2 1/4] ASoC: fsl_ssi: Switch to SPDX identifier
by Fabio Estevam 03 May '18

03 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Retain the copyright statements in fsl_ssi.h (Timur) sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 6 ++--- sound/soc/fsl/fsl_ssi_dbg.c | 18 ++++++-------- 3 files changed, 35 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1544166..0a64822 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1,34 +1,29 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * - * Some notes why imx-pcm-fiq is used instead of DMA on some boards: - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver +// +// Author: Timur Tabi <timur(a)freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// Some notes why imx-pcm-fiq is used instead of DMA on some boards: +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/init.h> #include <linux/io.h> diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 18f8dd5..0bdda60 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2008 Freescale Semiconductor, Inc. */ #ifndef _MPC8610_I2S_H diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e..1bacfa2 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -1,14 +1,10 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions - * - * Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix - * - * Splitted from fsl_ssi.c - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions +// +// Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix +// +// Split from fsl_ssi.c #include <linux/debugfs.h> #include <linux/device.h> -- 2.7.4

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