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
  ...

1 files changed, 0 insertions, 151 deletions

diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
deleted file mode 100644
index 9579799ace54..000000000000
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- * afeb9260.c  --  SoC audio for AFEB9260
- *
- * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-
-#include <linux/atmel-ssc.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-
-#include "../codecs/tlv320aic23.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-#define CODEC_CLOCK 	12000000
-
-static int afeb9260_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int err;
-
-	/* Set the codec system clock for DAC and ADC */
-	err =
-	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
-
-	if (err < 0) {
-		printk(KERN_ERR "can't set codec system clock\n");
-		return err;
-	}
-
-	return err;
-}
-
-static struct snd_soc_ops afeb9260_ops = {
-	.hw_params = afeb9260_hw_params,
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Jack", NULL),
-	SND_SOC_DAPM_LINE("Line In", NULL),
-	SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
-	{"Headphone Jack", NULL, "LHPOUT"},
-	{"Headphone Jack", NULL, "RHPOUT"},
-
-	{"LLINEIN", NULL, "Line In"},
-	{"RLINEIN", NULL, "Line In"},
-
-	{"MICIN", NULL, "Mic Jack"},
-};
-
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link afeb9260_dai = {
-	.name = "TLV320AIC23",
-	.stream_name = "AIC23",
-	.cpu_dai_name = "atmel-ssc-dai.0",
-	.codec_dai_name = "tlv320aic23-hifi",
-	.platform_name = "atmel_pcm-audio",
-	.codec_name = "tlv320aic23-codec.0-001a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
-		   SND_SOC_DAIFMT_CBM_CFM,
-	.ops = &afeb9260_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_machine_afeb9260 = {
-	.name = "AFEB9260",
-	.owner = THIS_MODULE,
-	.dai_link = &afeb9260_dai,
-	.num_links = 1,
-
-	.dapm_widgets = tlv320aic23_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
-	.dapm_routes = afeb9260_audio_map,
-	.num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
-};
-
-static struct platform_device *afeb9260_snd_device;
-
-static int __init afeb9260_soc_init(void)
-{
-	int err;
-	struct device *dev;
-
-	if (!(machine_is_afeb9260()))
-		return -ENODEV;
-
-
-	afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!afeb9260_snd_device) {
-		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
-		return -ENOMEM;
-	}
-
-	platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
-	err = platform_device_add(afeb9260_snd_device);
-	if (err)
-		goto err1;
-
-	dev = &afeb9260_snd_device->dev;
-
-	return 0;
-err1:
-	platform_device_put(afeb9260_snd_device);
-	return err;
-}
-
-static void __exit afeb9260_soc_exit(void)
-{
-	platform_device_unregister(afeb9260_snd_device);
-}
-
-module_init(afeb9260_soc_init);
-module_exit(afeb9260_soc_exit);
-
-MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
-MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
-MODULE_LICENSE("GPL");
-