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Commit 163ae6f3 authored by Takashi Sakamoto's avatar Takashi Sakamoto Committed by Takashi Iwai
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ALSA: firewire-digi00x: add data block processing layer



Digi 002/003 family uses its own format for data blocks. The format is
quite similar to AM824 in IEC 61883-6, while there're some differences:

 * The Valid Bit Length (VBL) code is always 0x40 in Multi-bit Linear Audio
   (MBLA) data channel.
 * The first data channel includes MIDI messages, against IEC 61883-6
   recommendation.
 * The Counter field is always zero in MIDI conformant data channel.
 * Sequence multiplexing in IEC 61883-6 is not applied to the MIDI
   conformant data channel.
 * PCM samples are scrambled in received AMDTP packets. We call the way
   as Double-Oh-Three (DOT). The algorithm was discovered by
   Robin Gareus and Damien Zammit in 2012.

This commit adds data processing layer to satisfy these differences.

There's a quirk about transmission mode for received packets. When this
driver applies non-blocking mode to outgoing packets with isochronous
channel 2 or more, after 15 to 20 seconds since playbacking, any PCM
samples causes noisy sound on the device. With isochronous channel 0 or 1,
this doesn't occur. As long as I investigated, this quirk is not observed
when applying blocking mode to the received packets.

This driver applies blocking mode to outgoing packets, while non-blocking
mode to incoming packgets.

Signed-off-by: default avatarTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent 9edf723f
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+1 −1
Original line number Diff line number Diff line
snd-firewire-digi00x-objs := digi00x.o
snd-firewire-digi00x-objs := amdtp-dot.o digi00x.o
obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
+330 −0
Original line number Diff line number Diff line
/*
 * 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

/*
 * 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;

	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, unsigned int midi_ports)
{
	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;
	p->midi_ports = midi_ports;

	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;
	}
}

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;
	}
}

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;
	}

	/* A place holder for MIDI processing. */

	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;
	}

	/* A place holder for MIDI processing. */

	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;
}
+14 −0
Original line number Diff line number Diff line
@@ -19,8 +19,12 @@

#include <sound/core.h>
#include <sound/initval.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>

#include "../lib.h"
#include "../iso-resources.h"
#include "../amdtp-stream.h"

struct snd_dg00x {
	struct snd_card *card;
@@ -29,4 +33,14 @@ struct snd_dg00x {
	struct mutex mutex;
};

int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
		   enum amdtp_stream_direction dir);
int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
			     unsigned int pcm_channels,
			     unsigned int midi_ports);
void amdtp_dot_reset(struct amdtp_stream *s);
int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
				     struct snd_pcm_runtime *runtime);
void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);

#endif