Merge tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We had again a busy development cycle with many new drivers as well as
  lots of core improvements / cleanups. Let's go for highlights:

  ALSA core:

   - PCM locking scheme was refactored for reducing a global rwlock

   - PCM suspend is handled in the device type PM ops now; lots of
     explicit calls were reduced by this action

   - Cleanups about PCM buffer preallocation calls

   - Kill NULL device object in memory allocations

   - Lots of procfs API cleanups

  ASoC core:

   - Support for only powering up channels that are actively being used

   - Cleanups / fixes of topology API

  ASoC drivers:

   - MediaTek BTCVSD for a Bluetooth radio chip, which is the first such
     driver we've had upstream!

   - Quite a few improvements to simplify the generic card drivers,
     especially the merge of the SCU cards into the main generic drivers

   - Lots of fixes for probing on Intel systems to follow more standard
     styles

   - A big refresh and cleanup of the Samsung drivers

   - New drivers: Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341
     and CS35L26, Google ChromeOS embedded controllers, Ingenic JZ4725B,
     MediaTek BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328,
     Spreadtrum DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM
     formatters

  ALSA drivers:

   - Improvements of Tegra HD-audio controller driver for supporting new
     chips

   - HD-audio codec quirks for ALC294 S4 resume, ASUS laptop, Chrome
     headset button support and Dell workstations

   - Improved DSD support on USB-audio

   - Quirk for MOTU MicroBook II USB-audio

   - Support for Fireface UCX support and Solid State Logic Duende
     Classic/Mini"

* tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (461 commits)
  ALSA: usb-audio: Add quirk for MOTU MicroBook II
  ASoC: stm32: i2s: skip useless write in slave mode
  ASoC: stm32: i2s: fix race condition in irq handler
  ASoC: stm32: i2s: remove useless callback
  ASoC: stm32: i2s: fix dma configuration
  ASoC: stm32: i2s: fix stream count management
  ASoC: stm32: i2s: fix 16 bit format support
  ASoC: stm32: i2s: fix IRQ clearing
  ASoC: qcom: Kconfig: fix dependency for sdm845
  ASoC: Intel: Boards: Add Maxim98373 support
  ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
  ALSA: firewire-motu: fix construction of PCM frame for capture direction
  ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
  ALSA: hda: Extend i915 component bind timeout
  ASoC: wm_adsp: Improve logging messages
  ASoC: wm_adsp: Add support for multiple compressed buffers
  ASoC: wm_adsp: Refactor compress stream initialisation
  ASoC: wm_adsp: Reorder some functions for improved clarity
  ASoC: wm_adsp: Factor out stripping padding from ADSP data
  ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug
  ...

3 files changed, 50 insertions, 6 deletions

diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 74244d8e2909..b2e9454f5816 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -376,7 +376,7 @@ void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus)
 {
 	unsigned long timeout;
 
-	snd_hdac_chip_updateb(bus, GCTL, 0, AZX_GCTL_RESET);
+	snd_hdac_chip_updateb(bus, GCTL, AZX_GCTL_RESET, AZX_GCTL_RESET);
 
 	timeout = jiffies + msecs_to_jiffies(100);
 	while (!snd_hdac_chip_readb(bus, GCTL) && time_before(jiffies, timeout))
@@ -415,7 +415,7 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset)
 	}
 
 	/* Accept unsolicited responses */
-	snd_hdac_chip_updatel(bus, GCTL, 0, AZX_GCTL_UNSOL);
+	snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, AZX_GCTL_UNSOL);
 
 	/* detect codecs */
 	if (!bus->codec_mask) {
@@ -431,7 +431,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_reset_link);
 static void azx_int_enable(struct hdac_bus *bus)
 {
 	/* enable controller CIE and GIE */
-	snd_hdac_chip_updatel(bus, INTCTL, 0, AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN);
+	snd_hdac_chip_updatel(bus, INTCTL,
+			      AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN,
+			      AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN);
 }
 
 /* disable interrupts */
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 617ff1aa818f..27eb0270a711 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -144,9 +144,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 		return -ENODEV;
 	if (!acomp->ops) {
 		request_module("i915");
-		/* 10s timeout */
+		/* 60s timeout */
 		wait_for_completion_timeout(&bind_complete,
-					    msecs_to_jiffies(10 * 1000));
+					    msecs_to_jiffies(60 * 1000));
 	}
 	if (!acomp->ops) {
 		dev_info(bus->dev, "couldn't bind with audio component\n");
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index eee422390d8e..f5dd288d1a7a 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -13,6 +13,40 @@
 #include "trace.h"
 
 /**
+ * snd_hdac_get_stream_stripe_ctl - get stripe control value
+ * @bus: HD-audio core bus
+ * @substream: PCM substream
+ */
+int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
+				   struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int channels = runtime->channels,
+		     rate = runtime->rate,
+		     bits_per_sample = runtime->sample_bits,
+		     max_sdo_lines, value, sdo_line;
+
+	/* T_AZA_GCAP_NSDO is 1:2 bitfields in GCAP */
+	max_sdo_lines = snd_hdac_chip_readl(bus, GCAP) & AZX_GCAP_NSDO;
+
+	/* following is from HD audio spec */
+	for (sdo_line = max_sdo_lines; sdo_line > 0; sdo_line >>= 1) {
+		if (rate > 48000)
+			value = (channels * bits_per_sample *
+					(rate / 48000)) / sdo_line;
+		else
+			value = (channels * bits_per_sample) / sdo_line;
+
+		if (value >= 8)
+			break;
+	}
+
+	/* stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines */
+	return sdo_line >> 1;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_get_stream_stripe_ctl);
+
+/**
  * snd_hdac_stream_init - initialize each stream (aka device)
  * @bus: HD-audio core bus
  * @azx_dev: HD-audio core stream object to initialize
@@ -48,6 +82,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_init);
 void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start)
 {
 	struct hdac_bus *bus = azx_dev->bus;
+	int stripe_ctl;
 
 	trace_snd_hdac_stream_start(bus, azx_dev);
 
@@ -56,7 +91,13 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start)
 		azx_dev->start_wallclk -= azx_dev->period_wallclk;
 
 	/* enable SIE */
-	snd_hdac_chip_updatel(bus, INTCTL, 0, 1 << azx_dev->index);
+	snd_hdac_chip_updatel(bus, INTCTL,
+			      1 << azx_dev->index,
+			      1 << azx_dev->index);
+	/* set stripe control */
+	stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream);
+	snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK,
+				stripe_ctl);
 	/* set DMA start and interrupt mask */
 	snd_hdac_stream_updateb(azx_dev, SD_CTL,
 				0, SD_CTL_DMA_START | SD_INT_MASK);
@@ -73,6 +114,7 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev)
 	snd_hdac_stream_updateb(azx_dev, SD_CTL,
 				SD_CTL_DMA_START | SD_INT_MASK, 0);
 	snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+	snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
 	azx_dev->running = false;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_stream_clear);