From: Takashi Sakamoto Date: Wed, 30 Sep 2015 00:39:17 +0000 (+0900) Subject: ALSA: firewire-digi00x: add data block processing layer X-Git-Url: https://git.stricted.de/?a=commitdiff_plain;h=163ae6f3f3f059ab44311792af5a73f70f747263;p=GitHub%2Fmoto-9609%2Fandroid_kernel_motorola_exynos9610.git 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 Signed-off-by: Takashi Iwai --- diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile index 13fec5bd4a81..87c4cfd2e8b0 100644 --- a/sound/firewire/digi00x/Makefile +++ b/sound/firewire/digi00x/Makefile @@ -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 diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c new file mode 100644 index 000000000000..e6731d33c480 --- /dev/null +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -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 + * Copyright (C) 2012 Damien Zammit + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#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; +} diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 0bb2ca3ae471..fe21d7778004 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -19,8 +19,12 @@ #include #include +#include +#include #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