Merge tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are no too intrusive changes in this update batch.  The biggest
  LOC is found in the new DICE driver, and other small changes are
  scattered over the whole sound subtree (which is a common pattern).

  Below are highlights:

   - ALSA core:
     * Memory allocation support with genpool
     * Fix blocking in drain ioctl of compress_offload

   - HD-audio:
     * Improved AMD HDMI supports
     * Intel HDMI detection improvements
     * thinkpad_acpi mute-key integration
     * New PCI ID, New ALC255,285,293 codecs, CX20952

   - USB-audio:
     * New buffer size management
     * Clean up endpoint handling codes

   - ASoC:
     * Further work on the dmaengine helpers, including support for
       configuring the parameters for DMA by reading the capabilities of
       the DMA controller which removes some guesswork and magic numbers
       from drivers.
     * A refresh of the documentation.
     * Conversions of many drivers to direct regmap API usage in order
       to allow the ASoC level register I/O code to be removed, this
       will hopefully be completed by v3.14.
     * Support for using async register I/O in DAPM, reducing the time
       taken to implement power transitions on systems that support it.

   - Firewire: DICE driver

   - Lots of small fixes for bugs reported by Coverity"

* tag 'sound-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (382 commits)
  ALSA: hda/realtek - Add new codec ALC255/ALC3234 UAJ supported
  ALSA: hda - Apply MacBook fixups for CS4208 correctly
  ASoC: fsl: imx-wm8962: remove an unneeded check
  ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable
  ALSA: hda/realtek - Make fixup regs persist after resume
  ALSA: hda_intel: ratelimit "spurious response" message
  ASoC: generic-dmaengine-pcm: Use SNDRV_DMA_TYPE_DEV_IRAM as default
  ASoC: dapm: Use WARN_ON() instead of BUG_ON()
  ASoC: wm_adsp: Fix BUG_ON() and WARN_ON() usages
  ASoC: Replace BUG() with WARN()
  ASoC: wm_hubs: Replace BUG() with WARN()
  ASoC: wm8996: Replace BUG() with WARN()
  ASoC: wm8962: Replace BUG() with WARN()
  ASoC: wm8958: Replace BUG() with WARN()
  ASoC: wm8904: Replace BUG() with WARN()
  ASoC: wm8900: Replace BUG() with WARN()
  ASoC: wm8350: Replace BUG() with WARN()
  ASoC: txx9: Use WARN_ON() instead of BUG_ON()
  ASoC: sh: Use WARN_ON() instead of BUG_ON()
  ASoC: rcar: Use WARN_ON() instead of BUG_ON()
  ...

1 files changed, 21 insertions, 25 deletions

diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index f6103d68c4b1..839ebf812d79 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -1,6 +1,7 @@
 #ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED
 #define SOUND_FIREWIRE_AMDTP_H_INCLUDED
 
+#include <linux/err.h>
 #include <linux/interrupt.h>
 #include <linux/mutex.h>
 #include "packets-buffer.h"
@@ -11,9 +12,18 @@
  *	sample_rate/8000 samples, with rounding up or down to adjust
  *	for clock skew and left-over fractional samples.  This should
  *	be used if supported by the device.
+ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or
+ *	SYT_INTERVAL samples, with these two types alternating so that
+ *	the overall sample rate comes out right.
+ * @CIP_HI_DUALWIRE: At rates above 96 kHz, pretend that the stream runs
+ *	at half the actual sample rate with twice the number of channels;
+ *	two samples of a channel are stored consecutively in the packet.
+ *	Requires blocking mode and SYT_INTERVAL-aligned PCM buffer size.
  */
 enum cip_out_flags {
-	CIP_NONBLOCKING = 0,
+	CIP_NONBLOCKING	= 0x00,
+	CIP_BLOCKING	= 0x01,
+	CIP_HI_DUALWIRE	= 0x02,
 };
 
 /**
@@ -27,6 +37,7 @@ enum cip_sfc {
 	CIP_SFC_96000  = 4,
 	CIP_SFC_176400 = 5,
 	CIP_SFC_192000 = 6,
+	CIP_SFC_COUNT
 };
 
 #define AMDTP_OUT_PCM_FORMAT_BITS	(SNDRV_PCM_FMTBIT_S16 | \
@@ -43,6 +54,7 @@ struct amdtp_out_stream {
 	struct mutex mutex;
 
 	enum cip_sfc sfc;
+	bool dual_wire;
 	unsigned int data_block_quadlets;
 	unsigned int pcm_channels;
 	unsigned int midi_ports;
@@ -51,6 +63,7 @@ struct amdtp_out_stream {
 				 __be32 *buffer, unsigned int frames);
 
 	unsigned int syt_interval;
+	unsigned int transfer_delay;
 	unsigned int source_node_id_field;
 	struct iso_packets_buffer buffer;
 
@@ -74,7 +87,10 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
 			  enum cip_out_flags flags);
 void amdtp_out_stream_destroy(struct amdtp_out_stream *s);
 
-void amdtp_out_stream_set_rate(struct amdtp_out_stream *s, unsigned int rate);
+void amdtp_out_stream_set_parameters(struct amdtp_out_stream *s,
+				     unsigned int rate,
+				     unsigned int pcm_channels,
+				     unsigned int midi_ports);
 unsigned int amdtp_out_stream_get_max_payload(struct amdtp_out_stream *s);
 
 int amdtp_out_stream_start(struct amdtp_out_stream *s, int channel, int speed);
@@ -87,31 +103,11 @@ void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s);
 unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s);
 void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
 
-/**
- * amdtp_out_stream_set_pcm - configure format of PCM samples
- * @s: the AMDTP output stream to be configured
- * @pcm_channels: the number of PCM samples in each data block, to be encoded
- *                as AM824 multi-bit linear audio
- *
- * This function must not be called while the stream is running.
- */
-static inline void amdtp_out_stream_set_pcm(struct amdtp_out_stream *s,
-					    unsigned int pcm_channels)
-{
-	s->pcm_channels = pcm_channels;
-}
+extern const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT];
 
-/**
- * amdtp_out_stream_set_midi - configure format of MIDI data
- * @s: the AMDTP output stream to be configured
- * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
- *
- * This function must not be called while the stream is running.
- */
-static inline void amdtp_out_stream_set_midi(struct amdtp_out_stream *s,
-					     unsigned int midi_ports)
+static inline bool amdtp_out_stream_running(struct amdtp_out_stream *s)
 {
-	s->midi_ports = midi_ports;
+	return !IS_ERR(s->context);
 }
 
 /**