Merge tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Here is the first batch of updates for sound system on 4.4-rc1.

  Again at this time, the update looks fairly calm; no big changes in
  either ALSA core or ASoC infrastructures, rather all small cleanups,
  in addition to the new stuff as usual.

  The biggest changes are about Firewire sound devices.  It gained lots
  of new device support, and MIDI functionality.  Also there are updates
  for a few still working-in-progress stuff (topology API and ASoC
  skylake), too.  But overall, this update should give no big surprise.

  Some highlights are below:

  Core:
   - A few more Kconfig items for tinification; it's marked as EXPERT,
     so normal user should't be bothered :)
   - Refactoring with a new PCM hw_constraint helper
   - Removal of unused transfer_ack_{begin,end} PCM callbacks

  Firewire:
   - Restructuring of code subtree, lots of refactoring
   - Support AMDTP variants
   - New driver for Digidesign 002/003 family
   - Adds support for TASCAM FireOne to ALSA OXFW driver
   - Add MIDI support to TASCAM and Digi00x devices

  HD-Audio:
   - Automated modalias generation for codec drivers, finally
   - Improvement on heuristics for setting mixer name
   - A few fixes for longstanding bugs on Creative CA0132 cards
   - Addition of audio rate callback with i915 communication
   - Fix suspend issue on recent Dell XPS
   - Intel Lewisburg controller support

  ASoC:
   - Updates to the topology userspace interface
   - Big updates to the Renesas support (rcar)
   - More updates for supporting Intel Sky Lake systems
   - New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10,
     Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825, Rockchip
     S/PDIF, and Atmel class D amplifier

  USB-Audio:
   - A fix for newer Roland MIDI devices
   - Quirks and workarounds for Zoom R16/24 device

  Misc:
   - A few fixes for some old Cirrus CS46xx PCI sound boards
   - Yet another fixes for some old ESS Maestro3 PCI sound boards"

* tag 'sound-4.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (330 commits)
  ALSA: hda - Add Intel Lewisburg device IDs Audio
  ALSA: hda - Apply pin fixup for HP ProBook 6550b
  ALSA: hda - Fix lost 4k BDL boundary workaround
  ALSA: maestro3: Fix Allegro mute until master volume/mute is touched
  ALSA: maestro3: Enable docking support for Dell Latitude C810
  ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports
  ALSA: firewire-digi00x: add MIDI operations for MIDI control port
  ALSA: firewire-digi00x: rename identifiers of MIDI operation for physical ports
  ALSA: cs46xx: Fix suspend for all channels
  ALSA: cs46xx: Fix Duplicate front for CS4294 and CS4298 codecs
  ALSA: DocBook: Add soc-ops.c and soc-compress.c
  ALSA: hda - Add / fix kernel doc comments
  ALSA: Constify ratden/ratnum constraints
  ALSA: hda - Disable 64bit address for Creative HDA controllers
  ALSA: hda/realtek - Dell XPS one ALC3260 speaker no sound after resume back
  ALSA: hda/ca0132 - Convert leftover pr_info() and pr_err()
  ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP
  ASoC: rt5645: Sort the order for register bit defines
  ASoC: dwc: add check for master/slave format
  ASoC: rt5645: Add the HWEQ for the speaker output
  ...

1 files changed, 442 insertions, 0 deletions

diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
new file mode 100644
index 000000000000..b02a5e8cad44
--- /dev/null
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -0,0 +1,442 @@
+/*
+ * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ * Copyright (C) 2012 Robin Gareus <robin@gareus.org>
+ * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "digi00x.h"
+
+#define CIP_FMT_AM		0x10
+
+/* 'Clock-based rate control mode' is just supported. */
+#define AMDTP_FDF_AM824		0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND	3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS	8
+
+/*
+ * The double-oh-three algorithm was discovered by Robin Gareus and Damien
+ * Zammit in 2012, with reverse-engineering for Digi 003 Rack.
+ */
+struct dot_state {
+	u8 carry;
+	u8 idx;
+	unsigned int off;
+};
+
+struct amdtp_dot {
+	unsigned int pcm_channels;
+	struct dot_state state;
+
+	unsigned int midi_ports;
+	/* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */
+	struct snd_rawmidi_substream *midi[2];
+	int midi_fifo_used[2];
+	int midi_fifo_limit;
+
+	void (*transfer_samples)(struct amdtp_stream *s,
+				 struct snd_pcm_substream *pcm,
+				 __be32 *buffer, unsigned int frames);
+};
+
+/*
+ * double-oh-three look up table
+ *
+ * @param idx index byte (audio-sample data) 0x00..0xff
+ * @param off channel offset shift
+ * @return salt to XOR with given data
+ */
+#define BYTE_PER_SAMPLE (4)
+#define MAGIC_DOT_BYTE (2)
+#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE)
+static const u8 dot_scrt(const u8 idx, const unsigned int off)
+{
+	/*
+	 * the length of the added pattern only depends on the lower nibble
+	 * of the last non-zero data
+	 */
+	static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14,
+				   12, 10, 8, 6, 4, 2, 0};
+
+	/*
+	 * the lower nibble of the salt. Interleaved sequence.
+	 * this is walked backwards according to len[]
+	 */
+	static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4,
+				   0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf};
+
+	/* circular list for the salt's hi nibble. */
+	static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4,
+				   0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa};
+
+	/*
+	 * start offset for upper nibble mapping.
+	 * note: 9 is /special/. In the case where the high nibble == 0x9,
+	 * hir[] is not used and - coincidentally - the salt's hi nibble is
+	 * 0x09 regardless of the offset.
+	 */
+	static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4,
+				   3, 0x00, 14, 13, 8, 9, 10, 2};
+
+	const u8 ln = idx & 0xf;
+	const u8 hn = (idx >> 4) & 0xf;
+	const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15];
+
+	if (len[ln] < off)
+		return 0x00;
+
+	return ((nib[14 + off - len[ln]]) | (hr << 4));
+}
+
+static void dot_encode_step(struct dot_state *state, __be32 *const buffer)
+{
+	u8 * const data = (u8 *) buffer;
+
+	if (data[MAGIC_DOT_BYTE] != 0x00) {
+		state->off = 0;
+		state->idx = data[MAGIC_DOT_BYTE] ^ state->carry;
+	}
+	data[MAGIC_DOT_BYTE] ^= state->carry;
+	state->carry = dot_scrt(state->idx, ++(state->off));
+}
+
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+			     unsigned int pcm_channels)
+{
+	struct amdtp_dot *p = s->protocol;
+	int err;
+
+	if (amdtp_stream_running(s))
+		return -EBUSY;
+
+	/*
+	 * A first data channel is for MIDI conformant data channel, the rest is
+	 * Multi Bit Linear Audio data channel.
+	 */
+	err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1);
+	if (err < 0)
+		return err;
+
+	s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+	p->pcm_channels = pcm_channels;
+
+	if (s->direction == AMDTP_IN_STREAM)
+		p->midi_ports = DOT_MIDI_IN_PORTS;
+	else
+		p->midi_ports = DOT_MIDI_OUT_PORTS;
+
+	/*
+	 * We do not know the actual MIDI FIFO size of most devices.  Just
+	 * assume two bytes, i.e., one byte can be received over the bus while
+	 * the previous one is transmitted over MIDI.
+	 * (The value here is adjusted for midi_ratelimit_per_packet().)
+	 */
+	p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+	return 0;
+}
+
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u32 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000);
+			dot_encode_step(&p->state, &buffer[c]);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u16 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
+			dot_encode_step(&p->state, &buffer[c]);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	u32 *dst;
+
+	channels = p->pcm_channels;
+	dst  = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			*dst = be32_to_cpu(buffer[c]) << 8;
+			dst++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			dst = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+			      unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int channels, i, c;
+
+	channels = p->pcm_channels;
+
+	buffer++;
+	for (i = 0; i < data_blocks; ++i) {
+		for (c = 0; c < channels; ++c)
+			buffer[c] = cpu_to_be32(0x40000000);
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+	struct amdtp_dot *p = s->protocol;
+	int used;
+
+	used = p->midi_fifo_used[port];
+	if (used == 0)
+		return true;
+
+	used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+	used = max(used, 0);
+	p->midi_fifo_used[port] = used;
+
+	return used < p->midi_fifo_limit;
+}
+
+static inline void midi_use_bytes(struct amdtp_stream *s,
+				  unsigned int port, unsigned int count)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count;
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+				unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int f, port;
+	int len;
+	u8 *b;
+
+	for (f = 0; f < data_blocks; f++) {
+		port = (s->data_block_counter + f) % 8;
+		b = (u8 *)&buffer[0];
+
+		len = 0;
+		if (port < p->midi_ports &&
+		    midi_ratelimit_per_packet(s, port) &&
+		    p->midi[port] != NULL)
+			len = snd_rawmidi_transmit(p->midi[port], b + 1, 2);
+
+		if (len > 0) {
+			b[3] = (0x10 << port) | len;
+			midi_use_bytes(s, port, len);
+		} else {
+			b[1] = 0;
+			b[2] = 0;
+			b[3] = 0;
+		}
+		b[0] = 0x80;
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+			       unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int f, port, len;
+	u8 *b;
+
+	for (f = 0; f < data_blocks; f++) {
+		b = (u8 *)&buffer[0];
+		port = b[3] >> 4;
+		len = b[3] & 0x0f;
+
+		if (port < p->midi_ports && p->midi[port] && len > 0)
+			snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+				     struct snd_pcm_runtime *runtime)
+{
+	int err;
+
+	/* This protocol delivers 24 bit data in 32bit data channel. */
+	err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+	if (err < 0)
+		return err;
+
+	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	if (WARN_ON(amdtp_stream_pcm_running(s)))
+		return;
+
+	switch (format) {
+	default:
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S16:
+		if (s->direction == AMDTP_OUT_STREAM) {
+			p->transfer_samples = write_pcm_s16;
+			break;
+		}
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S32:
+		if (s->direction == AMDTP_OUT_STREAM)
+			p->transfer_samples = write_pcm_s32;
+		else
+			p->transfer_samples = read_pcm_s32;
+		break;
+	}
+}
+
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+			  struct snd_rawmidi_substream *midi)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	if (port < p->midi_ports)
+		ACCESS_ONCE(p->midi[port]) = midi;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks;
+	} else {
+		pcm_frames = 0;
+	}
+
+	read_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks;
+	} else {
+		write_pcm_silence(s, buffer, data_blocks);
+		pcm_frames = 0;
+	}
+
+	write_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+		 enum amdtp_stream_direction dir)
+{
+	amdtp_stream_process_data_blocks_t process_data_blocks;
+	enum cip_flags flags;
+
+	/* Use different mode between incoming/outgoing. */
+	if (dir == AMDTP_IN_STREAM) {
+		flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+		process_data_blocks = process_tx_data_blocks;
+	} else {
+		flags = CIP_BLOCKING;
+		process_data_blocks = process_rx_data_blocks;
+	}
+
+	return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+				 process_data_blocks, sizeof(struct amdtp_dot));
+}
+
+void amdtp_dot_reset(struct amdtp_stream *s)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	p->state.carry = 0x00;
+	p->state.idx = 0x00;
+	p->state.off = 0;
+}