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: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Takashi Sakamoto 2015-09-30 09:39:17 +09:00 committed by Takashi Iwai
parent 9edf723fd8
commit 163ae6f3f3
3 changed files with 345 additions and 1 deletions

View file

@ -1,2 +1,2 @@
snd-firewire-digi00x-objs := digi00x.o
snd-firewire-digi00x-objs := amdtp-dot.o digi00x.o
obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o

View file

@ -0,0 +1,330 @@
/*
* 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;
}

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@ -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