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author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-12-11 13:20:50 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-12-11 13:20:50 -0800 |
commit | bae41e45b7400496b9bf0c70c6004419d9987819 (patch) | |
tree | cf22a65d119da1c414dbc79518857800fbe7a24b /sound/soc/pxa | |
parent | 7ef58b32f571bffb7763c6252ad7527562081f34 (diff) | |
parent | 6e1d7a51392f06899bd7b693f28ac60fa1e00032 (diff) | |
download | linux-bae41e45b7400496b9bf0c70c6004419d9987819.tar.bz2 |
Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
ALSA core:
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
USB-audio:
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
FireWire:
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
HD-audio:
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
ASoC:
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
Others:
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle"
* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
ALSA: pcxhr: NULL dereference on probe failure
ALSA: lola: NULL dereference on probe failure
ALSA: hda - Add "eapd" model string for AD1986A codec
ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
ALSA: oxfw: Add hwdep interface
ALSA: oxfw: Add support for capture/playback MIDI messages
ALSA: oxfw: add support for capturing PCM samples
ALSA: oxfw: Add support AMDTP in-stream
ALSA: oxfw: Add support for Behringer/Mackie devices
ALSA: oxfw: Change the way to start stream
ALSA: oxfw: Add proc interface for debugging purpose
ALSA: oxfw: Change the way to make PCM rules/constraints
ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
ALSA: oxfw: Change the way to name card
ALSA: dice: Add support for MIDI capture/playback
ALSA: dice: Add support for capturing PCM samples
ALSA: dice: Support for non SYT-Match sampling clock source mode
ALSA: dice: Add support for duplex streams with synchronization
ALSA: dice: Change the way to start stream
ALSA: jack: Add dummy snd_jack_set_key() definition
...
Diffstat (limited to 'sound/soc/pxa')
-rw-r--r-- | sound/soc/pxa/mioa701_wm9713.c | 7 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 16 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-ac97.c | 6 | ||||
-rw-r--r-- | sound/soc/pxa/spitz.c | 52 |
4 files changed, 41 insertions, 40 deletions
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 595eee341e90..a6b2be20cc0b 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -127,15 +127,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - unsigned short reg; /* Prepare GPIO8 for rear speaker amplifier */ - reg = codec->driver->read(codec, AC97_GPIO_CFG); - codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100); + snd_soc_update_bits(codec, AC97_GPIO_CFG, 0x100, 0x100); /* Prepare MIC input */ - reg = codec->driver->read(codec, AC97_3D_CONTROL); - codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000); + snd_soc_update_bits(codec, AC97_3D_CONTROL, 0xc000, 0xc000); return 0; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a8e097433074..cbba063a7210 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -97,7 +97,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); pxa_ssp_disable(ssp); } @@ -121,7 +121,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); } kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); @@ -136,7 +136,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); @@ -144,7 +144,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->psp = __raw_readl(ssp->mmio_base + SSPSP); pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -154,7 +154,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (cpu_dai->active) pxa_ssp_enable(ssp); else - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -256,11 +256,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (ssp->type != PXA3xx_SSP) - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); if (ssp->type != PXA3xx_SSP) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index ae956e3f4b9d..73ca2820c08c 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -157,7 +157,7 @@ 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", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -174,7 +174,7 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-aux", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -191,7 +191,7 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-mic", - .ac97_control = 1, + .bus_control = true, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1373b017a951..d7d5fb20ea6f 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -305,19 +305,15 @@ static struct snd_soc_card snd_soc_spitz = { .num_dapm_routes = ARRAY_SIZE(spitz_audio_map), }; -static struct platform_device *spitz_snd_device; - -static int __init spitz_init(void) +static int spitz_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) - return -ENODEV; - - if (machine_is_borzoi() || machine_is_spitz()) - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - else + if (machine_is_akita()) spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; + else + spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); if (ret) @@ -327,37 +323,45 @@ static int __init spitz_init(void) if (ret) goto err2; - spitz_snd_device = platform_device_alloc("soc-audio", -1); - if (!spitz_snd_device) { - ret = -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); goto err2; } - platform_set_drvdata(spitz_snd_device, &snd_soc_spitz); - - ret = platform_device_add(spitz_snd_device); - if (ret) - goto err3; - return 0; -err3: - platform_device_put(spitz_snd_device); err2: gpio_free(spitz_mic_gpio); err1: return ret; } -static void __exit spitz_exit(void) +static int spitz_remove(struct platform_device *pdev) { - platform_device_unregister(spitz_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); gpio_free(spitz_mic_gpio); + return 0; } -module_init(spitz_init); -module_exit(spitz_exit); +static struct platform_driver spitz_driver = { + .driver = { + .name = "spitz-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = spitz_probe, + .remove = spitz_remove, +}; + +module_platform_driver(spitz_driver); MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Spitz"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spitz-audio"); |