Merge tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There've been many updates in ASoC side at this time, especially the
  framework enhancement for multiple CODECs on a single DAI and more
  componentization works.

  The only major change in ALSA core is the addition of timestamp type
  in sw_params field.  This should behave in backward compatible way.

  Other than that, there are lots of small changes and new drivers in
  wide range, including a large code cut in HD-audio driver for
  deprecated static quirks.  Some highlights are below:

  ALSA Core:
   - Add the new timestamp type field to sw_params to choose
     MONOTONIC_RAW type

  HD-audio:
   - Continued conversion to standard printk macros, generic code
     cleanups
   - Removal of obsoleted static quirk codes for Conexant and C-Media
     codecs
   - Fixups for HP Envy TS, Dell XPS 15, HP and Dell mute/mic LED,
     Gigabyte BXBT-2807 mobo
   - Intel Braswell support

  ASoC:
   - Support for multiple CODECs attached to a single DAI, enabling
     systems with for example multiple DAC/speaker drivers on a single
     link, contributed by Benoit Cousson based on work from Misael Lopez
     Cruz
   - Support for byte controls larger than 256 bytes based on the use of
     TLVs contributed by Omair Mohammed Abdullah
   - More componentisation work from Lars-Peter Clausen
   - The remainder of the conversions of CODEC drivers to params_width()
     by Mark Brown
   - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks,
     Realtek RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas
     Instruments TAS2552
   - Lots of updates and fixes, especially to the DaVinci, Intel,
     Freescale, Realtek, and rcar drivers"

* tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (402 commits)
  ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*
  ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI input
  sound/oss/pss: Remove typedefs pss_mixerdata and pss_confdata
  sound/oss/opl3: Remove typedef opl_devinfo
  ALSA: fireworks: fix specifiers in format strings for propper output
  ASoC: imx-audmux: Use uintptr_t for port numbers
  ASoC: davinci: Enable menuconfig entry for McASP
  ASoC: fsl_asrc: Don't access members of config before checking it
  ASoC: fsl_sarc_dma: Check pair before using it
  ASoC: adau1977: Fix truncation warning on 64 bit architectures
  ALSA: virtuoso: add Xonar Essence STX II support
  ALSA: riptide: fix %d confusingly prefixed with 0x in format strings
  ALSA: fireworks: fix %d confusingly prefixed with 0x in format strings
  ALSA: hda - add codec ID for Braswell display audio codec
  ALSA: hda - add PCI IDs for Intel Braswell
  ALSA: usb-audio: Adjust Gamecom 780 volume level
  ALSA: usb-audio: improve dmesg source grepability
  ASoC: rt5670: Fix duplicate const warnings
  ASoC: rt5670: Staticise non-exported symbols
  ASoC: Intel: update stream only on stream IPC msgs
  ...

1 files changed, 251 insertions, 0 deletions

diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
new file mode 100644
index 000000000000..0e550f14028f
--- /dev/null
+++ b/sound/soc/intel/broadwell.c
@@ -0,0 +1,251 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC1", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphones", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+	{"LINE1", NULL, "Line Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC1"},
+	{"DMIC2 Pin", NULL, "DMIC2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int broadwell_rt286_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 ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+	.hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	/* always connected - check HP for jack detect */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line Jack");
+	snd_soc_dapm_enable_pin(dapm, "DMIC1");
+	snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = broadwell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload0",
+		.stream_name = "Offload0 Playback",
+		.cpu_dai_name = "Offload0 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload1",
+		.stream_name = "Offload1 Playback",
+		.cpu_dai_name = "Offload1 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Loopback PCM",
+		.stream_name = "Loopback",
+		.cpu_dai_name = "Loopback Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 0,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Capture PCM",
+		.stream_name = "Capture",
+		.cpu_dai_name = "Capture Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.be_id = 0,
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &broadwell_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+	.name = "broadwell-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = broadwell_rt286_dais,
+	.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+	.dapm_widgets = broadwell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+	.dapm_routes = broadwell_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+	.fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+	broadwell_rt286.dev = &pdev->dev;
+
+	return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&broadwell_rt286);
+	return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+	.probe = broadwell_audio_probe,
+	.remove = broadwell_audio_remove,
+	.driver = {
+		.name = "broadwell-audio",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");