// SPDX-License-Identifier: GPL-2.0-only /* * bebob_stream.c - a part of driver for BeBoB based devices * * Copyright (c) 2013-2014 Takashi Sakamoto */ #include "./bebob.h" #define CALLBACK_TIMEOUT 2500 #define FW_ISO_RESOURCE_DELAY 1000 /* * NOTE; * For BeBoB streams, Both of input and output CMP connection are important. * * For most devices, each CMP connection starts to transmit/receive a * corresponding stream. But for a few devices, both of CMP connection needs * to start transmitting stream. An example is 'M-Audio Firewire 410'. */ /* 128 is an arbitrary length but it seems to be enough */ #define FORMAT_MAXIMUM_LENGTH 128 const unsigned int snd_bebob_rate_table[SND_BEBOB_STRM_FMT_ENTRIES] = { [0] = 32000, [1] = 44100, [2] = 48000, [3] = 88200, [4] = 96000, [5] = 176400, [6] = 192000, }; /* * See: Table 51: Extended Stream Format Info ‘Sampling Frequency’ * in Additional AVC commands (Nov 2003, BridgeCo) */ static const unsigned int bridgeco_freq_table[] = { [0] = 0x02, [1] = 0x03, [2] = 0x04, [3] = 0x0a, [4] = 0x05, [5] = 0x06, [6] = 0x07, }; static int get_formation_index(unsigned int rate, unsigned int *index) { unsigned int i; for (i = 0; i < ARRAY_SIZE(snd_bebob_rate_table); i++) { if (snd_bebob_rate_table[i] == rate) { *index = i; return 0; } } return -EINVAL; } int snd_bebob_stream_get_rate(struct snd_bebob *bebob, unsigned int *curr_rate) { unsigned int tx_rate, rx_rate, trials; int err; trials = 0; do { err = avc_general_get_sig_fmt(bebob->unit, &tx_rate, AVC_GENERAL_PLUG_DIR_OUT, 0); } while (err == -EAGAIN && ++trials < 3); if (err < 0) goto end; trials = 0; do { err = avc_general_get_sig_fmt(bebob->unit, &rx_rate, AVC_GENERAL_PLUG_DIR_IN, 0); } while (err == -EAGAIN && ++trials < 3); if (err < 0) goto end; *curr_rate = rx_rate; if (rx_rate == tx_rate) goto end; /* synchronize receive stream rate to transmit stream rate */ err = avc_general_set_sig_fmt(bebob->unit, rx_rate, AVC_GENERAL_PLUG_DIR_IN, 0); end: return err; } int snd_bebob_stream_set_rate(struct snd_bebob *bebob, unsigned int rate) { int err; err = avc_general_set_sig_fmt(bebob->unit, rate, AVC_GENERAL_PLUG_DIR_OUT, 0); if (err < 0) goto end; err = avc_general_set_sig_fmt(bebob->unit, rate, AVC_GENERAL_PLUG_DIR_IN, 0); if (err < 0) goto end; /* * Some devices need a bit time for transition. * 300msec is got by some experiments. */ msleep(300); end: return err; } int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, enum snd_bebob_clock_type *src) { const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7]; unsigned int id; enum avc_bridgeco_plug_type type; int err = 0; /* 1.The device has its own operation to switch source of clock */ if (clk_spec) { err = clk_spec->get(bebob, &id); if (err < 0) { dev_err(&bebob->unit->device, "fail to get clock source: %d\n", err); goto end; } if (id >= clk_spec->num) { dev_err(&bebob->unit->device, "clock source %d out of range 0..%d\n", id, clk_spec->num - 1); err = -EIO; goto end; } *src = clk_spec->types[id]; goto end; } /* * 2.The device don't support to switch source of clock then assumed * to use internal clock always */ if (bebob->sync_input_plug < 0) { *src = SND_BEBOB_CLOCK_TYPE_INTERNAL; goto end; } /* * 3.The device supports to switch source of clock by an usual way. * Let's check input for 'Music Sub Unit Sync Input' plug. */ avc_bridgeco_fill_msu_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, bebob->sync_input_plug); err = avc_bridgeco_get_plug_input(bebob->unit, addr, input); if (err < 0) { dev_err(&bebob->unit->device, "fail to get an input for MSU in plug %d: %d\n", bebob->sync_input_plug, err); goto end; } /* * If there are no input plugs, all of fields are 0xff. * Here check the first field. This field is used for direction. */ if (input[0] == 0xff) { *src = SND_BEBOB_CLOCK_TYPE_INTERNAL; goto end; } /* The source from any output plugs is for one purpose only. */ if (input[0] == AVC_BRIDGECO_PLUG_DIR_OUT) { /* * In BeBoB architecture, the source from music subunit may * bypass from oPCR[0]. This means that this source gives * synchronization to IEEE 1394 cycle start packet. */ if (input[1] == AVC_BRIDGECO_PLUG_MODE_SUBUNIT && input[2] == 0x0c) { *src = SND_BEBOB_CLOCK_TYPE_INTERNAL; goto end; } /* The source from any input units is for several purposes. */ } else if (input[1] == AVC_BRIDGECO_PLUG_MODE_UNIT) { if (input[2] == AVC_BRIDGECO_PLUG_UNIT_ISOC) { if (input[3] == 0x00) { /* * This source comes from iPCR[0]. This means * that presentation timestamp calculated by * SYT series of the received packets. In * short, this driver is the master of * synchronization. */ *src = SND_BEBOB_CLOCK_TYPE_SYT; goto end; } else { /* * This source comes from iPCR[1-29]. This * means that the synchronization stream is not * the Audio/MIDI compound stream. */ *src = SND_BEBOB_CLOCK_TYPE_EXTERNAL; goto end; } } else if (input[2] == AVC_BRIDGECO_PLUG_UNIT_EXT) { /* Check type of this plug. */ avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, AVC_BRIDGECO_PLUG_UNIT_EXT, input[3]); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) goto end; if (type == AVC_BRIDGECO_PLUG_TYPE_DIG) { /* * SPDIF/ADAT or sometimes (not always) word * clock. */ *src = SND_BEBOB_CLOCK_TYPE_EXTERNAL; goto end; } else if (type == AVC_BRIDGECO_PLUG_TYPE_SYNC) { /* Often word clock. */ *src = SND_BEBOB_CLOCK_TYPE_EXTERNAL; goto end; } else if (type == AVC_BRIDGECO_PLUG_TYPE_ADDITION) { /* * Not standard. * Mostly, additional internal clock. */ *src = SND_BEBOB_CLOCK_TYPE_INTERNAL; goto end; } } } /* Not supported. */ err = -EIO; end: return err; } static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; unsigned int stm_pos, sec_loc, pos; u8 *buf, addr[AVC_BRIDGECO_ADDR_BYTES], type; enum avc_bridgeco_plug_dir dir; int err; /* * The length of return value of this command cannot be expected. Here * use the maximum length of FCP. */ buf = kzalloc(256, GFP_KERNEL); if (buf == NULL) return -ENOMEM; if (s == &bebob->tx_stream) dir = AVC_BRIDGECO_PLUG_DIR_OUT; else dir = AVC_BRIDGECO_PLUG_DIR_IN; avc_bridgeco_fill_unit_addr(addr, dir, AVC_BRIDGECO_PLUG_UNIT_ISOC, 0); err = avc_bridgeco_get_plug_ch_pos(bebob->unit, addr, buf, 256); if (err < 0) { dev_err(&bebob->unit->device, "fail to get channel position for isoc %s plug 0: %d\n", (dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" : "out", err); goto end; } pos = 0; /* positions in I/O buffer */ pcm = 0; midi = 0; /* the number of sections in AMDTP packet */ sections = buf[pos++]; for (sec = 0; sec < sections; sec++) { /* type of this section */ avc_bridgeco_fill_unit_addr(addr, dir, AVC_BRIDGECO_PLUG_UNIT_ISOC, 0); err = avc_bridgeco_get_plug_section_type(bebob->unit, addr, sec, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get section type for isoc %s plug 0: %d\n", (dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" : "out", err); goto end; } /* NoType */ if (type == 0xff) { err = -ENOSYS; goto end; } /* the number of channels in this section */ channels = buf[pos++]; for (ch = 0; ch < channels; ch++) { /* position of this channel in AMDTP packet */ stm_pos = buf[pos++] - 1; /* location of this channel in this section */ sec_loc = buf[pos++] - 1; /* * Basically the number of location is within the * number of channels in this section. But some models * of M-Audio don't follow this. Its location for MIDI * is the position of MIDI channels in AMDTP packet. */ if (sec_loc >= channels) sec_loc = ch; switch (type) { /* for MIDI conformant data channel */ case 0x0a: /* AMDTP_MAX_CHANNELS_FOR_MIDI is 1. */ if ((midi > 0) && (stm_pos != midi)) { err = -ENOSYS; goto end; } amdtp_am824_set_midi_position(s, stm_pos); midi = stm_pos; break; /* for PCM data channel */ case 0x01: /* Headphone */ case 0x02: /* Microphone */ case 0x03: /* Line */ case 0x04: /* SPDIF */ case 0x05: /* ADAT */ case 0x06: /* TDIF */ case 0x07: /* MADI */ /* for undefined/changeable signal */ case 0x08: /* Analog */ case 0x09: /* Digital */ default: location = pcm + sec_loc; if (location >= AM824_MAX_CHANNELS_FOR_PCM) { err = -ENOSYS; goto end; } amdtp_am824_set_pcm_position(s, location, stm_pos); break; } } if (type != 0x0a) pcm += channels; else midi += channels; } end: kfree(buf); return err; } static int check_connection_used_by_others(struct snd_bebob *bebob, struct amdtp_stream *s) { struct cmp_connection *conn; bool used; int err; if (s == &bebob->tx_stream) conn = &bebob->out_conn; else conn = &bebob->in_conn; err = cmp_connection_check_used(conn, &used); if ((err >= 0) && used && !amdtp_stream_running(s)) { dev_err(&bebob->unit->device, "Connection established by others: %cPCR[%d]\n", (conn->direction == CMP_OUTPUT) ? 'o' : 'i', conn->pcr_index); err = -EBUSY; } return err; } static void break_both_connections(struct snd_bebob *bebob) { cmp_connection_break(&bebob->in_conn); cmp_connection_break(&bebob->out_conn); // These models seem to be in transition state for a longer time. When // accessing in the state, any transactions is corrupted. In the worst // case, the device is going to reboot. if (bebob->version < 2) msleep(600); } static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) { struct cmp_connection *conn; int err = 0; if (stream == &bebob->rx_stream) conn = &bebob->in_conn; else conn = &bebob->out_conn; // channel mapping. if (bebob->maudio_special_quirk == NULL) { err = map_data_channels(bebob, stream); if (err < 0) return err; } err = cmp_connection_establish(conn); if (err < 0) return err; return amdtp_domain_add_stream(&bebob->domain, stream, conn->resources.channel, conn->speed); } static int init_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) { enum amdtp_stream_direction dir_stream; struct cmp_connection *conn; enum cmp_direction dir_conn; int err; if (stream == &bebob->tx_stream) { dir_stream = AMDTP_IN_STREAM; conn = &bebob->out_conn; dir_conn = CMP_OUTPUT; } else { dir_stream = AMDTP_OUT_STREAM; conn = &bebob->in_conn; dir_conn = CMP_INPUT; } err = cmp_connection_init(conn, bebob->unit, dir_conn, 0); if (err < 0) return err; err = amdtp_am824_init(stream, bebob->unit, dir_stream, CIP_BLOCKING); if (err < 0) { cmp_connection_destroy(conn); return err; } if (stream == &bebob->tx_stream) { // BeBoB v3 transfers packets with these qurks: // - In the beginning of streaming, the value of dbc is // incremented even if no data blocks are transferred. // - The value of dbc is reset suddenly. if (bebob->version > 2) bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC | CIP_SKIP_DBC_ZERO_CHECK; // At high sampling rate, M-Audio special firmware transmits // empty packet with the value of dbc incremented by 8 but the // others are valid to IEC 61883-1. if (bebob->maudio_special_quirk) bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC; } return 0; } static void destroy_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) { amdtp_stream_destroy(stream); if (stream == &bebob->tx_stream) cmp_connection_destroy(&bebob->out_conn); else cmp_connection_destroy(&bebob->in_conn); } int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) { int err; err = init_stream(bebob, &bebob->tx_stream); if (err < 0) return err; err = init_stream(bebob, &bebob->rx_stream); if (err < 0) { destroy_stream(bebob, &bebob->tx_stream); return err; } err = amdtp_domain_init(&bebob->domain); if (err < 0) { destroy_stream(bebob, &bebob->tx_stream); destroy_stream(bebob, &bebob->rx_stream); } return err; } static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { unsigned int pcm_channels; unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { pcm_channels = bebob->tx_stream_formations[index].pcm; midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { pcm_channels = bebob->rx_stream_formations[index].pcm; midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; return cmp_connection_reserve(conn, amdtp_stream_get_max_payload(stream)); } int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, unsigned int frames_per_period, unsigned int frames_per_buffer) { unsigned int curr_rate; int err; // Considering JACK/FFADO streaming: // TODO: This can be removed hwdep functionality becomes popular. err = check_connection_used_by_others(bebob, &bebob->rx_stream); if (err < 0) return err; err = bebob->spec->rate->get(bebob, &curr_rate); if (err < 0) return err; if (rate == 0) rate = curr_rate; if (curr_rate != rate) { amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); cmp_connection_release(&bebob->out_conn); cmp_connection_release(&bebob->in_conn); } if (bebob->substreams_counter == 0 || curr_rate != rate) { unsigned int index; // NOTE: // If establishing connections at first, Yamaha GO46 // (and maybe Terratec X24) don't generate sound. // // For firmware customized by M-Audio, refer to next NOTE. err = bebob->spec->rate->set(bebob, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to set sampling rate: %d\n", err); return err; } err = get_formation_index(rate, &index); if (err < 0) return err; err = keep_resources(bebob, &bebob->tx_stream, rate, index); if (err < 0) return err; err = keep_resources(bebob, &bebob->rx_stream, rate, index); if (err < 0) { cmp_connection_release(&bebob->out_conn); return err; } err = amdtp_domain_set_events_per_period(&bebob->domain, frames_per_period, frames_per_buffer); if (err < 0) { cmp_connection_release(&bebob->out_conn); cmp_connection_release(&bebob->in_conn); return err; } } return 0; } int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) { int err; // Need no substreams. if (bebob->substreams_counter == 0) return -EIO; // packet queueing error or detecting discontinuity if (amdtp_streaming_error(&bebob->rx_stream) || amdtp_streaming_error(&bebob->tx_stream)) { amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); } if (!amdtp_stream_running(&bebob->rx_stream)) { enum snd_bebob_clock_type src; struct amdtp_stream *master, *slave; unsigned int curr_rate; unsigned int ir_delay_cycle; if (bebob->maudio_special_quirk) { err = bebob->spec->rate->get(bebob, &curr_rate); if (err < 0) return err; } err = snd_bebob_stream_get_clock_src(bebob, &src); if (err < 0) return err; if (src != SND_BEBOB_CLOCK_TYPE_SYT) { master = &bebob->tx_stream; slave = &bebob->rx_stream; } else { master = &bebob->rx_stream; slave = &bebob->tx_stream; } err = start_stream(bebob, master); if (err < 0) goto error; err = start_stream(bebob, slave); if (err < 0) goto error; // The device postpones start of transmission mostly for 1 sec // after receives packets firstly. For safe, IR context starts // 0.4 sec (=3200 cycles) later to version 1 or 2 firmware, // 2.0 sec (=16000 cycles) for version 3 firmware. This is // within 2.5 sec (=CALLBACK_TIMEOUT). // Furthermore, some devices transfer isoc packets with // discontinuous counter in the beginning of packet streaming. // The delay has an effect to avoid detection of this // discontinuity. if (bebob->version < 2) ir_delay_cycle = 3200; else ir_delay_cycle = 16000; err = amdtp_domain_start(&bebob->domain, ir_delay_cycle); if (err < 0) goto error; // NOTE: // The firmware customized by M-Audio uses these commands to // start transmitting stream. This is not usual way. if (bebob->maudio_special_quirk) { err = bebob->spec->rate->set(bebob, curr_rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to ensure sampling rate: %d\n", err); goto error; } } if (!amdtp_stream_wait_callback(&bebob->rx_stream, CALLBACK_TIMEOUT) || !amdtp_stream_wait_callback(&bebob->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } } return 0; error: amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); return err; } void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { if (bebob->substreams_counter == 0) { amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); cmp_connection_release(&bebob->out_conn); cmp_connection_release(&bebob->in_conn); } } /* * This function should be called before starting streams or after stopping * streams. */ void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob) { amdtp_domain_destroy(&bebob->domain); destroy_stream(bebob, &bebob->tx_stream); destroy_stream(bebob, &bebob->rx_stream); } /* * See: Table 50: Extended Stream Format Info Format Hierarchy Level 2’ * in Additional AVC commands (Nov 2003, BridgeCo) * Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005 */ static int parse_stream_formation(u8 *buf, unsigned int len, struct snd_bebob_stream_formation *formation) { unsigned int i, e, channels, format; /* * this module can support a hierarchy combination that: * Root: Audio and Music (0x90) * Level 1: AM824 Compound (0x40) */ if ((buf[0] != 0x90) || (buf[1] != 0x40)) return -ENOSYS; /* check sampling rate */ for (i = 0; i < ARRAY_SIZE(bridgeco_freq_table); i++) { if (buf[2] == bridgeco_freq_table[i]) break; } if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */ memset(&formation[i], 0, sizeof(struct snd_bebob_stream_formation)); for (e = 0; e < buf[4]; e++) { channels = buf[5 + e * 2]; format = buf[6 + e * 2]; switch (format) { /* IEC 60958 Conformant, currently handled as MBLA */ case 0x00: /* Multi bit linear audio */ case 0x06: /* Raw */ formation[i].pcm += channels; break; /* MIDI Conformant */ case 0x0d: formation[i].midi += channels; break; /* IEC 61937-3 to 7 */ case 0x01: case 0x02: case 0x03: case 0x04: case 0x05: /* Multi bit linear audio */ case 0x07: /* DVD-Audio */ case 0x0c: /* High Precision */ /* One Bit Audio */ case 0x08: /* (Plain) Raw */ case 0x09: /* (Plain) SACD */ case 0x0a: /* (Encoded) Raw */ case 0x0b: /* (Encoded) SACD */ /* Synchronization Stream (Stereo Raw audio) */ case 0x40: /* Don't care */ case 0xff: default: return -ENOSYS; /* not supported */ } } if (formation[i].pcm > AM824_MAX_CHANNELS_FOR_PCM || formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI) return -ENOSYS; return 0; } static int fill_stream_formations(struct snd_bebob *bebob, enum avc_bridgeco_plug_dir dir, unsigned short pid) { u8 *buf; struct snd_bebob_stream_formation *formations; unsigned int len, eid; u8 addr[AVC_BRIDGECO_ADDR_BYTES]; int err; buf = kmalloc(FORMAT_MAXIMUM_LENGTH, GFP_KERNEL); if (buf == NULL) return -ENOMEM; if (dir == AVC_BRIDGECO_PLUG_DIR_IN) formations = bebob->rx_stream_formations; else formations = bebob->tx_stream_formations; for (eid = 0; eid < SND_BEBOB_STRM_FMT_ENTRIES; eid++) { len = FORMAT_MAXIMUM_LENGTH; avc_bridgeco_fill_unit_addr(addr, dir, AVC_BRIDGECO_PLUG_UNIT_ISOC, pid); err = avc_bridgeco_get_plug_strm_fmt(bebob->unit, addr, buf, &len, eid); /* No entries remained. */ if (err == -EINVAL && eid > 0) { err = 0; break; } else if (err < 0) { dev_err(&bebob->unit->device, "fail to get stream format %d for isoc %s plug %d:%d\n", eid, (dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" : "out", pid, err); break; } err = parse_stream_formation(buf, len, formations); if (err < 0) break; } kfree(buf); return err; } static int seek_msu_sync_input_plug(struct snd_bebob *bebob) { u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES]; unsigned int i; enum avc_bridgeco_plug_type type; int err; /* Get the number of Music Sub Unit for both direction. */ err = avc_general_get_plug_info(bebob->unit, 0x0c, 0x00, 0x00, plugs); if (err < 0) { dev_err(&bebob->unit->device, "fail to get info for MSU in/out plugs: %d\n", err); goto end; } /* seek destination plugs for 'MSU sync input' */ bebob->sync_input_plug = -1; for (i = 0; i < plugs[0]; i++) { avc_bridgeco_fill_msu_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, i); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get type for MSU in plug %d: %d\n", i, err); goto end; } if (type == AVC_BRIDGECO_PLUG_TYPE_SYNC) { bebob->sync_input_plug = i; break; } } end: return err; } int snd_bebob_stream_discover(struct snd_bebob *bebob) { const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES]; enum avc_bridgeco_plug_type type; unsigned int i; int err; /* the number of plugs for isoc in/out, ext in/out */ err = avc_general_get_plug_info(bebob->unit, 0x1f, 0x07, 0x00, plugs); if (err < 0) { dev_err(&bebob->unit->device, "fail to get info for isoc/external in/out plugs: %d\n", err); goto end; } /* * This module supports at least one isoc input plug and one isoc * output plug. */ if ((plugs[0] == 0) || (plugs[1] == 0)) { err = -ENOSYS; goto end; } avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, AVC_BRIDGECO_PLUG_UNIT_ISOC, 0); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get type for isoc in plug 0: %d\n", err); goto end; } else if (type != AVC_BRIDGECO_PLUG_TYPE_ISOC) { err = -ENOSYS; goto end; } err = fill_stream_formations(bebob, AVC_BRIDGECO_PLUG_DIR_IN, 0); if (err < 0) goto end; avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_OUT, AVC_BRIDGECO_PLUG_UNIT_ISOC, 0); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get type for isoc out plug 0: %d\n", err); goto end; } else if (type != AVC_BRIDGECO_PLUG_TYPE_ISOC) { err = -ENOSYS; goto end; } err = fill_stream_formations(bebob, AVC_BRIDGECO_PLUG_DIR_OUT, 0); if (err < 0) goto end; /* count external input plugs for MIDI */ bebob->midi_input_ports = 0; for (i = 0; i < plugs[2]; i++) { avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, AVC_BRIDGECO_PLUG_UNIT_EXT, i); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get type for external in plug %d: %d\n", i, err); goto end; } else if (type == AVC_BRIDGECO_PLUG_TYPE_MIDI) { bebob->midi_input_ports++; } } /* count external output plugs for MIDI */ bebob->midi_output_ports = 0; for (i = 0; i < plugs[3]; i++) { avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_OUT, AVC_BRIDGECO_PLUG_UNIT_EXT, i); err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type); if (err < 0) { dev_err(&bebob->unit->device, "fail to get type for external out plug %d: %d\n", i, err); goto end; } else if (type == AVC_BRIDGECO_PLUG_TYPE_MIDI) { bebob->midi_output_ports++; } } /* for check source of clock later */ if (!clk_spec) err = seek_msu_sync_input_plug(bebob); end: return err; } void snd_bebob_stream_lock_changed(struct snd_bebob *bebob) { bebob->dev_lock_changed = true; wake_up(&bebob->hwdep_wait); } int snd_bebob_stream_lock_try(struct snd_bebob *bebob) { int err; spin_lock_irq(&bebob->lock); /* user land lock this */ if (bebob->dev_lock_count < 0) { err = -EBUSY; goto end; } /* this is the first time */ if (bebob->dev_lock_count++ == 0) snd_bebob_stream_lock_changed(bebob); err = 0; end: spin_unlock_irq(&bebob->lock); return err; } void snd_bebob_stream_lock_release(struct snd_bebob *bebob) { spin_lock_irq(&bebob->lock); if (WARN_ON(bebob->dev_lock_count <= 0)) goto end; if (--bebob->dev_lock_count == 0) snd_bebob_stream_lock_changed(bebob); end: spin_unlock_irq(&bebob->lock); }