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author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-01-28 16:26:57 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-01-28 16:26:57 -0800 |
commit | fb95aae6e67c4e319a24b3eea32032d4246a5335 (patch) | |
tree | c310d68211634ef594d180fdd93844fec44de2fe /sound/soc/pxa | |
parent | bd2463ac7d7ec51d432f23bf0e893fb371a908cd (diff) | |
parent | 90fb04f890bcb7384b4d4c216dc2640b0a870df3 (diff) | |
download | linux-fb95aae6e67c4e319a24b3eea32032d4246a5335.tar.bz2 |
Merge tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"As the diffstat shows we've had again a lot of works done for this
cycle: the majority of changes are the continued componentization and
code refactoring in ASoC, the tree-wide PCM API updates and cleanups
and SOF updates while a few ASoC driver updates are seen, too.
Here we go, some highlights:
Core:
- Finally y2038 support landed to ALSA ABI; some ioctls have been
extended and lots of tricks were applied
- Applying the new managed PCM buffer API to all drivers; the API
itself was already merged in 5.5
- The already deprecated dimension support in ALSA control API is
dropped completely now
- Verification of ALSA control elements to catch API misuses
ASoC:
- Further code refactorings and moving things to the component level
- Lots of updates and improvements on SOF / Intel drivers; now
including common HDMI driver and SoundWire support
- New driver support for Ingenic JZ4770, Mediatek MT6660, Qualcomm
WCD934x and WSA881x, and Realtek RT700, RT711, RT715, RT1011,
RT1015 and RT1308
HD-audio:
- Improved ring-buffer communications using waitqueue
- Drop the superfluous buffer preallocation on x86
Others:
- Many code cleanups, mostly constifications over the whole tree
- USB-audio: quirks for MOTU, Corsair Virtuoso, Line6 Helix
- FireWire: code refactoring for oxfw and dice drivers"
* tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (638 commits)
ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
ALSA: hda: Add Clevo W65_67SB the power_save blacklist
ASoC: soc-core: remove null_snd_soc_ops
ASoC: soc-pcm: add soc_rtd_trigger()
ASoC: soc-pcm: add soc_rtd_hw_free()
ASoC: soc-pcm: add soc_rtd_hw_params()
ASoC: soc-pcm: add soc_rtd_prepare()
ASoC: soc-pcm: add soc_rtd_shutdown()
ASoC: soc-pcm: add soc_rtd_startup()
ASoC: rt1015: add rt1015 amplifier driver
ASoC: madera: Correct some kernel doc
ASoC: topology: fix soc_tplg_fe_link_create() - link->dobj initialization order
ASoC: Intel: skl_hda_dsp_common: Fix global-out-of-bounds bug
ASoC: madera: Correct DMIC only input hook ups
ALSA: cs46xx: fix spelling mistake "to" -> "too"
ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
ASoC: Add MediaTek MT6660 Speaker Amp Driver
ASoC: dt-bindings: rt5645: add suppliers
ASoC: max98090: fix deadlock in max98090_dapm_put_enum_double()
ASoC: dapm: add snd_soc_dapm_put_enum_double_locked
...
Diffstat (limited to 'sound/soc/pxa')
-rw-r--r-- | sound/soc/pxa/mioa701_wm9713.c | 2 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-pcm.c | 1 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 17 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-ac97.c | 4 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.c | 13 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-pcm.c | 1 |
6 files changed, 15 insertions, 23 deletions
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 129eb5251a5f..76e054d514a8 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -72,7 +72,7 @@ static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); component = rtd->codec_dai->component; return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event)); } diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 54a4c9213e83..287b5da739e5 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -225,7 +225,6 @@ static const struct snd_soc_component_driver mmp_soc_component = { .name = DRV_NAME, .open = mmp_pcm_open, .close = mmp_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = mmp_pcm_hw_params, .trigger = mmp_pcm_trigger, .pointer = mmp_pcm_pointer, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 76fdce54f007..e615acaa0199 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -133,12 +133,12 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM -static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) +static int pxa_ssp_suspend(struct snd_soc_component *component) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_priv *priv = snd_soc_component_get_drvdata(component); struct ssp_device *ssp = priv->ssp; - if (!cpu_dai->active) + if (!component->active) clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); @@ -151,9 +151,9 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) return 0; } -static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) +static int pxa_ssp_resume(struct snd_soc_component *component) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_priv *priv = snd_soc_component_get_drvdata(component); struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) __raw_writel(priv->to, ssp->mmio_base + SSTO); __raw_writel(priv->psp, ssp->mmio_base + SSPSP); - if (cpu_dai->active) + if (component->active) pxa_ssp_enable(ssp); else clk_disable_unprepare(ssp->clk); @@ -850,8 +850,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { static struct snd_soc_dai_driver pxa_ssp_dai = { .probe = pxa_ssp_probe, .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, .playback = { .channels_min = 1, .channels_max = 8, @@ -873,13 +871,14 @@ static const struct snd_soc_component_driver pxa_ssp_component = { .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = pxa2xx_soc_pcm_hw_params, .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, .trigger = pxa2xx_soc_pcm_trigger, .pointer = pxa2xx_soc_pcm_pointer, .mmap = pxa2xx_soc_pcm_mmap, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 31e81a6f616f..4240fde6aae8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -157,7 +157,6 @@ static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", - .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -174,7 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-aux", - .bus_control = true, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -191,7 +189,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-mic", - .bus_control = true, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, @@ -208,7 +205,6 @@ static const struct snd_soc_component_driver pxa_ac97_component = { .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = pxa2xx_soc_pcm_hw_params, .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e77d707efde7..5f1c477b5833 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -261,7 +261,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, } #ifdef CONFIG_PM -static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) +static int pxa2xx_soc_pcm_suspend(struct snd_soc_component *component) { /* store registers */ pxa_i2s.sacr0 = SACR0; @@ -275,7 +275,7 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) return 0; } -static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) +static int pxa2xx_soc_pcm_resume(struct snd_soc_component *component) { pxa_i2s_wait(); @@ -290,8 +290,8 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) } #else -#define pxa2xx_i2s_suspend NULL -#define pxa2xx_i2s_resume NULL +#define pxa2xx_soc_pcm_suspend NULL +#define pxa2xx_soc_pcm_resume NULL #endif static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) @@ -342,8 +342,6 @@ static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { static struct snd_soc_dai_driver pxa_i2s_dai = { .probe = pxa2xx_i2s_probe, .remove = pxa2xx_i2s_remove, - .suspend = pxa2xx_i2s_suspend, - .resume = pxa2xx_i2s_resume, .playback = { .channels_min = 2, .channels_max = 2, @@ -364,13 +362,14 @@ static const struct snd_soc_component_driver pxa_i2s_component = { .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = pxa2xx_soc_pcm_hw_params, .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, .trigger = pxa2xx_soc_pcm_trigger, .pointer = pxa2xx_soc_pcm_pointer, .mmap = pxa2xx_soc_pcm_mmap, + .suspend = pxa2xx_soc_pcm_suspend, + .resume = pxa2xx_soc_pcm_resume, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 07b3455a6f23..2b7839715dd5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -22,7 +22,6 @@ static const struct snd_soc_component_driver pxa2xx_soc_platform = { .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = pxa2xx_soc_pcm_hw_params, .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, |