diff options
Diffstat (limited to 'sound')
114 files changed, 5073 insertions, 1598 deletions
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7b977b753a03..7985dd8198b6 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, vendor_id); ret = device_add(&codec->dev); - if (ret) - goto err_free_codec; + if (ret) { + put_device(&codec->dev); + return ret; + } return 0; -err_free_codec: - of_node_put(codec->dev.of_node); - put_device(&codec->dev); - kfree(codec); - ac97_ctrl->codecs[idx] = NULL; - - return ret; } unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index db917546965d..9827bee109c1 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -71,8 +71,10 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) return 0; } v = i2c_smbus_read_byte_data(onyx->i2c, reg); - if (v < 0) + if (v < 0) { + *value = 0; return -1; + } *value = (u8)v; onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; return 0; diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99b882158705..41905afada63 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; @@ -705,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: return -EPERM; + default: + break; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { snd_compr_drain_notify(stream); @@ -795,9 +807,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); if (retval) { @@ -817,6 +837,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; + /* next track doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) + return -EPERM; + /* you can signal next track if this is intended to be a gapless stream * and current track metadata is set */ @@ -834,9 +858,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) static int snd_compr_partial_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + + /* partial drain doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) return -EPERM; + /* stream can be drained only when next track has been signalled */ if (stream->next_track == false) return -EPERM; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 34390be3fb0f..91c6ad58729f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group) spin_lock_init(&group->lock); mutex_init(&group->mutex); INIT_LIST_HEAD(&group->substreams); - refcount_set(&group->refs, 0); + refcount_set(&group->refs, 1); } /* define group lock helpers */ @@ -220,13 +220,12 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - /* architecture supports dma_mmap_coherent()? */ -#if defined(CONFIG_ARCH_NO_COHERENT_DMA_MMAP) || !defined(CONFIG_HAS_DMA) - if (!substream->ops->mmap && - substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return false; -#endif - return true; + + if (substream->ops->mmap || + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + return true; + + return dma_can_mmap(substream->dma_buffer.dev.dev); } static int constrain_mask_params(struct snd_pcm_substream *substream, @@ -1096,8 +1095,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group, if (!group) return; - do_free = refcount_dec_and_test(&group->refs) && - list_empty(&group->substreams); + do_free = refcount_dec_and_test(&group->refs); snd_pcm_group_unlock(group, substream->pcm->nonatomic); if (do_free) kfree(group); @@ -1874,6 +1872,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (!to_check) break; /* all drained */ init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); if (runtime->no_period_wakeup) @@ -1886,7 +1885,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } tout = msecs_to_jiffies(tout * 1000); } - tout = schedule_timeout_interruptible(tout); + tout = schedule_timeout(tout); snd_pcm_stream_lock_irq(substream); group = snd_pcm_stream_group_ref(substream); @@ -2020,6 +2019,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_lock_irq(target_group, nonatomic); snd_pcm_stream_lock(substream1); snd_pcm_group_assign(substream1, target_group); + refcount_inc(&target_group->refs); snd_pcm_stream_unlock(substream1); snd_pcm_group_unlock_irq(target_group, nonatomic); _end: @@ -2056,13 +2056,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) snd_pcm_group_lock_irq(group, nonatomic); relink_to_local(substream); + refcount_dec(&group->refs); /* detach the last stream, too */ if (list_is_singular(&group->substreams)) { relink_to_local(list_first_entry(&group->substreams, struct snd_pcm_substream, link_list)); - do_free = !refcount_read(&group->refs); + do_free = refcount_dec_and_test(&group->refs); } snd_pcm_group_unlock_irq(group, nonatomic); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7737b2670064..6d9592f0ae1d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1835,8 +1835,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr->type == USER_CLIENT) { info->input_pool = cptr->data.user.fifo_pool_size; info->input_free = info->input_pool; - if (cptr->data.user.fifo) - info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool); + info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo); } else { info->input_pool = 0; info->input_free = 0; diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index ea69261f269a..eaaa8b5830bb 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -263,3 +263,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return 0; } + +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) +{ + unsigned long flags; + int cells; + + if (!f) + return 0; + + snd_use_lock_use(&f->use_lock); + spin_lock_irqsave(&f->lock, flags); + cells = snd_seq_unused_cells(f->pool); + spin_unlock_irqrestore(&f->lock, flags); + snd_use_lock_free(&f->use_lock); + return cells; +} diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h index edc68743943d..b56a7b897c9c 100644 --- a/sound/core/seq/seq_fifo.h +++ b/sound/core/seq/seq_fifo.h @@ -53,5 +53,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table /* resize pool in fifo */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize); +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f); #endif diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index fd5d6b8ac557..67d735e9a6a4 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -146,19 +146,24 @@ void amdtp_am824_set_midi_position(struct amdtp_stream *s, } EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position); -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) +static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_am824 *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; const u32 *src; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { @@ -172,19 +177,24 @@ static void write_pcm_s32(struct amdtp_stream *s, } } -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) +static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_am824 *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; u32 *dst; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { @@ -284,7 +294,7 @@ static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port) } static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, - unsigned int frames) + unsigned int frames, unsigned int data_block_counter) { struct amdtp_am824 *p = s->protocol; unsigned int f, port; @@ -293,7 +303,7 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, for (f = 0; f < frames; f++) { b = (u8 *)&buffer[p->midi_position]; - port = (s->data_block_counter + f) % 8; + port = (data_block_counter + f) % 8; if (f < MAX_MIDI_RX_BLOCKS && midi_ratelimit_per_packet(s, port) && p->midi[port] != NULL && @@ -311,16 +321,20 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, } } -static void read_midi_messages(struct amdtp_stream *s, - __be32 *buffer, unsigned int frames) +static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer, + unsigned int frames, unsigned int data_block_counter) { struct amdtp_am824 *p = s->protocol; - unsigned int f, port; int len; u8 *b; + int f; for (f = 0; f < frames; f++) { - port = (8 - s->ctx_data.tx.first_dbc + s->data_block_counter + f) % 8; + unsigned int port = f; + + if (!(s->flags & CIP_UNALIGHED_DBC)) + port += data_block_counter; + port %= 8; b = (u8 *)&buffer[p->midi_position]; len = b[0] - 0x80; @@ -331,43 +345,60 @@ static void read_midi_messages(struct amdtp_stream *s, } } -static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer, - unsigned int data_blocks, unsigned int *syt) +static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { struct amdtp_am824 *p = s->protocol; - struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); - unsigned int pcm_frames; - - if (pcm) { - write_pcm_s32(s, pcm, buffer, data_blocks); - pcm_frames = data_blocks * p->frame_multiplier; - } else { - write_pcm_silence(s, buffer, data_blocks); - pcm_frames = 0; - } + unsigned int pcm_frames = 0; + int i; - if (p->midi_ports) - write_midi_messages(s, buffer, data_blocks); + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; + + if (pcm) { + write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks * p->frame_multiplier; + } else { + write_pcm_silence(s, buf, data_blocks); + } + + if (p->midi_ports) { + write_midi_messages(s, buf, data_blocks, + desc->data_block_counter); + } + } return pcm_frames; } -static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer, - unsigned int data_blocks, unsigned int *syt) +static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { struct amdtp_am824 *p = s->protocol; - struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); - unsigned int pcm_frames; - - if (pcm) { - read_pcm_s32(s, pcm, buffer, data_blocks); - pcm_frames = data_blocks * p->frame_multiplier; - } else { - pcm_frames = 0; - } + unsigned int pcm_frames = 0; + int i; + + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - if (p->midi_ports) - read_midi_messages(s, buffer, data_blocks); + if (pcm) { + read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks * p->frame_multiplier; + } + + if (p->midi_ports) { + read_midi_messages(s, buf, data_blocks, + desc->data_block_counter); + } + } return pcm_frames; } @@ -383,15 +414,14 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffe int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, enum cip_flags flags) { - amdtp_stream_process_data_blocks_t process_data_blocks; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; if (dir == AMDTP_IN_STREAM) - process_data_blocks = process_tx_data_blocks; + process_ctx_payloads = process_ir_ctx_payloads; else - process_data_blocks = process_rx_data_blocks; + process_ctx_payloads = process_it_ctx_payloads; return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM, - process_data_blocks, - sizeof(struct amdtp_am824)); + process_ctx_payloads, sizeof(struct amdtp_am824)); } EXPORT_SYMBOL_GPL(amdtp_am824_init); diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 4adbbf789cbe..16c7f6605511 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -14,8 +14,8 @@ #include <linux/tracepoint.h> TRACE_EVENT(amdtp_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int index), - TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, index), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int data_block_counter, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, data_block_counter, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -47,7 +47,7 @@ TRACE_EVENT(amdtp_packet, } __entry->payload_quadlets = payload_length / sizeof(__be32); __entry->data_blocks = data_blocks; - __entry->data_block_counter = s->data_block_counter, + __entry->data_block_counter = data_block_counter, __entry->packet_index = s->packet_index; __entry->irq = !!in_interrupt(); __entry->index = index; diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4d71d74707cf..e50e28f77e74 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -74,16 +74,16 @@ static void pcm_period_tasklet(unsigned long data); * @dir: the direction of stream * @flags: the packet transmission method to use * @fmt: the value of fmt field in CIP header - * @process_data_blocks: callback handler to process data blocks + * @process_ctx_payloads: callback handler to process payloads of isoc context * @protocol_size: the size to allocate newly for protocol */ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, enum cip_flags flags, unsigned int fmt, - amdtp_stream_process_data_blocks_t process_data_blocks, + amdtp_stream_process_ctx_payloads_t process_ctx_payloads, unsigned int protocol_size) { - if (process_data_blocks == NULL) + if (process_ctx_payloads == NULL) return -EINVAL; s->protocol = kzalloc(protocol_size, GFP_KERNEL); @@ -102,7 +102,10 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->fmt = fmt; - s->process_data_blocks = process_data_blocks; + s->process_ctx_payloads = process_ctx_payloads; + + if (dir == AMDTP_OUT_STREAM) + s->ctx_data.rx.syt_override = -1; return 0; } @@ -473,12 +476,12 @@ static inline int queue_in_packet(struct amdtp_stream *s, } static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2], - unsigned int syt) + unsigned int data_block_counter, unsigned int syt) { cip_header[0] = cpu_to_be32(READ_ONCE(s->source_node_id_field) | (s->data_block_quadlets << CIP_DBS_SHIFT) | ((s->sph << CIP_SPH_SHIFT) & CIP_SPH_MASK) | - s->data_block_counter); + data_block_counter); cip_header[1] = cpu_to_be32(CIP_EOH | ((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) | ((s->ctx_data.rx.fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) | @@ -487,8 +490,9 @@ static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2], static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle, struct fw_iso_packet *params, - unsigned int data_blocks, unsigned int syt, - unsigned int index) + unsigned int data_blocks, + unsigned int data_block_counter, + unsigned int syt, unsigned int index) { unsigned int payload_length; __be32 *cip_header; @@ -496,14 +500,9 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle, payload_length = data_blocks * sizeof(__be32) * s->data_block_quadlets; params->payload_length = payload_length; - if (s->flags & CIP_DBC_IS_END_EVENT) { - s->data_block_counter = - (s->data_block_counter + data_blocks) & 0xff; - } - if (!(s->flags & CIP_NO_HEADER)) { cip_header = (__be32 *)params->header; - generate_cip_header(s, cip_header, syt); + generate_cip_header(s, cip_header, data_block_counter, syt); params->header_length = 2 * sizeof(__be32); payload_length += params->header_length; } else { @@ -511,23 +510,19 @@ static void build_it_pkt_header(struct amdtp_stream *s, unsigned int cycle, } trace_amdtp_packet(s, cycle, cip_header, payload_length, data_blocks, - index); - - if (!(s->flags & CIP_DBC_IS_END_EVENT)) { - s->data_block_counter = - (s->data_block_counter + data_blocks) & 0xff; - } + data_block_counter, index); } static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, unsigned int payload_length, - unsigned int *data_blocks, unsigned int *dbc, - unsigned int *syt) + unsigned int *data_blocks, + unsigned int *data_block_counter, unsigned int *syt) { u32 cip_header[2]; unsigned int sph; unsigned int fmt; unsigned int fdf; + unsigned int dbc; bool lost; cip_header[0] = be32_to_cpu(buf[0]); @@ -579,17 +574,16 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } /* Check data block counter continuity */ - *dbc = cip_header[0] & CIP_DBC_MASK; + dbc = cip_header[0] & CIP_DBC_MASK; if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && - s->data_block_counter != UINT_MAX) - *dbc = s->data_block_counter; + *data_block_counter != UINT_MAX) + dbc = *data_block_counter; - if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) && - *dbc == s->ctx_data.tx.first_dbc) || - s->data_block_counter == UINT_MAX) { + if ((dbc == 0x00 && (s->flags & CIP_SKIP_DBC_ZERO_CHECK)) || + *data_block_counter == UINT_MAX) { lost = false; } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { - lost = *dbc != s->data_block_counter; + lost = dbc != *data_block_counter; } else { unsigned int dbc_interval; @@ -598,16 +592,18 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, else dbc_interval = *data_blocks; - lost = *dbc != ((s->data_block_counter + dbc_interval) & 0xff); + lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } if (lost) { dev_err(&s->unit->device, "Detect discontinuity of CIP: %02X %02X\n", - s->data_block_counter, *dbc); + *data_block_counter, dbc); return -EIO; } + *data_block_counter = dbc; + *syt = cip_header[1] & CIP_SYT_MASK; return 0; @@ -616,10 +612,10 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, const __be32 *ctx_header, unsigned int *payload_length, - unsigned int *data_blocks, unsigned int *syt, - unsigned int index) + unsigned int *data_blocks, + unsigned int *data_block_counter, + unsigned int *syt, unsigned int index) { - unsigned int dbc; const __be32 *cip_header; int err; @@ -635,7 +631,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, if (!(s->flags & CIP_NO_HEADER)) { cip_header = ctx_header + 2; err = check_cip_header(s, cip_header, *payload_length, - data_blocks, &dbc, syt); + data_blocks, data_block_counter, syt); if (err < 0) return err; } else { @@ -645,16 +641,12 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, s->data_block_quadlets; *syt = 0; - if (s->data_block_counter != UINT_MAX) - dbc = s->data_block_counter; - else - dbc = 0; + if (*data_block_counter == UINT_MAX) + *data_block_counter = 0; } - s->data_block_counter = dbc; - trace_amdtp_packet(s, cycle, cip_header, *payload_length, *data_blocks, - index); + *data_block_counter, index); return err; } @@ -686,6 +678,80 @@ static inline u32 compute_it_cycle(const __be32 ctx_header_tstamp) return increment_cycle_count(cycle, QUEUE_LENGTH); } +static int generate_device_pkt_descs(struct amdtp_stream *s, + struct pkt_desc *descs, + const __be32 *ctx_header, + unsigned int packets) +{ + unsigned int dbc = s->data_block_counter; + int i; + int err; + + for (i = 0; i < packets; ++i) { + struct pkt_desc *desc = descs + i; + unsigned int index = (s->packet_index + i) % QUEUE_LENGTH; + unsigned int cycle; + unsigned int payload_length; + unsigned int data_blocks; + unsigned int syt; + + cycle = compute_cycle_count(ctx_header[1]); + + err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length, + &data_blocks, &dbc, &syt, i); + if (err < 0) + return err; + + desc->cycle = cycle; + desc->syt = syt; + desc->data_blocks = data_blocks; + desc->data_block_counter = dbc; + desc->ctx_payload = s->buffer.packets[index].buffer; + + if (!(s->flags & CIP_DBC_IS_END_EVENT)) + dbc = (dbc + desc->data_blocks) & 0xff; + + ctx_header += + s->ctx_data.tx.ctx_header_size / sizeof(*ctx_header); + } + + s->data_block_counter = dbc; + + return 0; +} + +static void generate_ideal_pkt_descs(struct amdtp_stream *s, + struct pkt_desc *descs, + const __be32 *ctx_header, + unsigned int packets) +{ + unsigned int dbc = s->data_block_counter; + int i; + + for (i = 0; i < packets; ++i) { + struct pkt_desc *desc = descs + i; + unsigned int index = (s->packet_index + i) % QUEUE_LENGTH; + + desc->cycle = compute_it_cycle(*ctx_header); + desc->syt = calculate_syt(s, desc->cycle); + desc->data_blocks = calculate_data_blocks(s, desc->syt); + + if (s->flags & CIP_DBC_IS_END_EVENT) + dbc = (dbc + desc->data_blocks) & 0xff; + + desc->data_block_counter = dbc; + + if (!(s->flags & CIP_DBC_IS_END_EVENT)) + dbc = (dbc + desc->data_blocks) & 0xff; + + desc->ctx_payload = s->buffer.packets[index].buffer; + + ++ctx_header; + } + + s->data_block_counter = dbc; +} + static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; @@ -694,6 +760,19 @@ static inline void cancel_stream(struct amdtp_stream *s) WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } +static void process_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets) +{ + struct snd_pcm_substream *pcm; + unsigned int pcm_frames; + + pcm = READ_ONCE(s->pcm); + pcm_frames = s->process_ctx_payloads(s, descs, packets, pcm); + if (pcm) + update_pcm_pointers(s, pcm, pcm_frames); +} + static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) @@ -706,38 +785,31 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, if (s->packet_index < 0) return; + generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets); + + process_ctx_payloads(s, s->pkt_descs, packets); + for (i = 0; i < packets; ++i) { - u32 cycle; + const struct pkt_desc *desc = s->pkt_descs + i; unsigned int syt; - unsigned int data_blocks; - __be32 *buffer; - unsigned int pcm_frames; struct { struct fw_iso_packet params; __be32 header[IT_PKT_HEADER_SIZE_CIP / sizeof(__be32)]; } template = { {0}, {0} }; - struct snd_pcm_substream *pcm; - cycle = compute_it_cycle(*ctx_header); - syt = calculate_syt(s, cycle); - data_blocks = calculate_data_blocks(s, syt); - buffer = s->buffer.packets[s->packet_index].buffer; - pcm_frames = s->process_data_blocks(s, buffer, data_blocks, - &syt); + if (s->ctx_data.rx.syt_override < 0) + syt = desc->syt; + else + syt = s->ctx_data.rx.syt_override; - build_it_pkt_header(s, cycle, &template.params, data_blocks, + build_it_pkt_header(s, desc->cycle, &template.params, + desc->data_blocks, desc->data_block_counter, syt, i); if (queue_out_packet(s, &template.params) < 0) { cancel_stream(s); return; } - - pcm = READ_ONCE(s->pcm); - if (pcm && pcm_frames > 0) - update_pcm_pointers(s, pcm, pcm_frames); - - ++ctx_header; } fw_iso_context_queue_flush(s->context); @@ -748,8 +820,10 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int i, packets; + unsigned int packets; __be32 *ctx_header = header; + int i; + int err; if (s->packet_index < 0) return; @@ -757,48 +831,23 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, // The number of packets in buffer. packets = header_length / s->ctx_data.tx.ctx_header_size; - for (i = 0; i < packets; i++) { - u32 cycle; - unsigned int payload_length; - unsigned int data_blocks; - unsigned int syt; - __be32 *buffer; - unsigned int pcm_frames = 0; - struct fw_iso_packet params = {0}; - struct snd_pcm_substream *pcm; - int err; - - cycle = compute_cycle_count(ctx_header[1]); - err = parse_ir_ctx_header(s, cycle, ctx_header, &payload_length, - &data_blocks, &syt, i); - if (err < 0 && err != -EAGAIN) - break; - - if (err >= 0) { - buffer = s->buffer.packets[s->packet_index].buffer; - pcm_frames = s->process_data_blocks(s, buffer, - data_blocks, &syt); - - if (!(s->flags & CIP_DBC_IS_END_EVENT)) { - s->data_block_counter += data_blocks; - s->data_block_counter &= 0xff; - } + err = generate_device_pkt_descs(s, s->pkt_descs, ctx_header, packets); + if (err < 0) { + if (err != -EAGAIN) { + cancel_stream(s); + return; } - - if (queue_in_packet(s, ¶ms) < 0) - break; - - pcm = READ_ONCE(s->pcm); - if (pcm && pcm_frames > 0) - update_pcm_pointers(s, pcm, pcm_frames); - - ctx_header += s->ctx_data.tx.ctx_header_size / sizeof(*ctx_header); + } else { + process_ctx_payloads(s, s->pkt_descs, packets); } - /* Queueing error or detecting invalid payload. */ - if (i < packets) { - cancel_stream(s); - return; + for (i = 0; i < packets; ++i) { + struct fw_iso_packet params = {0}; + + if (queue_in_packet(s, ¶ms) < 0) { + cancel_stream(s); + return; + } } fw_iso_context_queue_flush(s->context); @@ -845,7 +894,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, * amdtp_stream_set_parameters() and it must be started before any PCM or MIDI * device can be started. */ -int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) +static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) { static const struct { unsigned int data_block; @@ -932,6 +981,13 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) else s->tag = TAG_CIP; + s->pkt_descs = kcalloc(INTERRUPT_INTERVAL, sizeof(*s->pkt_descs), + GFP_KERNEL); + if (!s->pkt_descs) { + err = -ENOMEM; + goto err_context; + } + s->packet_index = 0; do { struct fw_iso_packet params; @@ -943,7 +999,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) err = queue_out_packet(s, ¶ms); } if (err < 0) - goto err_context; + goto err_pkt_descs; } while (s->packet_index > 0); /* NOTE: TAG1 matches CIP. This just affects in stream. */ @@ -954,12 +1010,13 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) s->callbacked = false; err = fw_iso_context_start(s->context, -1, 0, tag); if (err < 0) - goto err_context; + goto err_pkt_descs; mutex_unlock(&s->mutex); return 0; - +err_pkt_descs: + kfree(s->pkt_descs); err_context: fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); @@ -970,7 +1027,6 @@ err_unlock: return err; } -EXPORT_SYMBOL(amdtp_stream_start); /** * amdtp_stream_pcm_pointer - get the PCM buffer position @@ -1041,7 +1097,7 @@ EXPORT_SYMBOL(amdtp_stream_update); * All PCM and MIDI devices of the stream must be stopped before the stream * itself can be stopped. */ -void amdtp_stream_stop(struct amdtp_stream *s) +static void amdtp_stream_stop(struct amdtp_stream *s) { mutex_lock(&s->mutex); @@ -1055,12 +1111,12 @@ void amdtp_stream_stop(struct amdtp_stream *s) fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); iso_packets_buffer_destroy(&s->buffer, s->unit); + kfree(s->pkt_descs); s->callbacked = false; mutex_unlock(&s->mutex); } -EXPORT_SYMBOL(amdtp_stream_stop); /** * amdtp_stream_pcm_abort - abort the running PCM device @@ -1078,3 +1134,92 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s) snd_pcm_stop_xrun(pcm); } EXPORT_SYMBOL(amdtp_stream_pcm_abort); + +/** + * amdtp_domain_init - initialize an AMDTP domain structure + * @d: the AMDTP domain to initialize. + */ +int amdtp_domain_init(struct amdtp_domain *d) +{ + INIT_LIST_HEAD(&d->streams); + + return 0; +} +EXPORT_SYMBOL_GPL(amdtp_domain_init); + +/** + * amdtp_domain_destroy - destroy an AMDTP domain structure + * @d: the AMDTP domain to destroy. + */ +void amdtp_domain_destroy(struct amdtp_domain *d) +{ + // At present nothing to do. + return; +} +EXPORT_SYMBOL_GPL(amdtp_domain_destroy); + +/** + * amdtp_domain_add_stream - register isoc context into the domain. + * @d: the AMDTP domain. + * @s: the AMDTP stream. + * @channel: the isochronous channel on the bus. + * @speed: firewire speed code. + */ +int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s, + int channel, int speed) +{ + struct amdtp_stream *tmp; + + list_for_each_entry(tmp, &d->streams, list) { + if (s == tmp) + return -EBUSY; + } + + list_add(&s->list, &d->streams); + + s->channel = channel; + s->speed = speed; + + return 0; +} +EXPORT_SYMBOL_GPL(amdtp_domain_add_stream); + +/** + * amdtp_domain_start - start sending packets for isoc context in the domain. + * @d: the AMDTP domain. + */ +int amdtp_domain_start(struct amdtp_domain *d) +{ + struct amdtp_stream *s; + int err = 0; + + list_for_each_entry(s, &d->streams, list) { + err = amdtp_stream_start(s, s->channel, s->speed); + if (err < 0) + break; + } + + if (err < 0) { + list_for_each_entry(s, &d->streams, list) + amdtp_stream_stop(s); + } + + return err; +} +EXPORT_SYMBOL_GPL(amdtp_domain_start); + +/** + * amdtp_domain_stop - stop sending packets for isoc context in the same domain. + * @d: the AMDTP domain to which the isoc contexts belong. + */ +void amdtp_domain_stop(struct amdtp_domain *d) +{ + struct amdtp_stream *s, *next; + + list_for_each_entry_safe(s, next, &d->streams, list) { + list_del(&s->list); + + amdtp_stream_stop(s); + } +} +EXPORT_SYMBOL_GPL(amdtp_domain_stop); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 3942894c11ac..bbbca964b9b4 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -33,6 +33,8 @@ * @CIP_HEADER_WITHOUT_EOH: Only for in-stream. CIP Header doesn't include * valid EOH. * @CIP_NO_HEADERS: a lack of headers in packets + * @CIP_UNALIGHED_DBC: Only for in-stream. The value of dbc is not alighed to + * the value of current SYT_INTERVAL; e.g. initial value is not zero. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -45,6 +47,7 @@ enum cip_flags { CIP_JUMBO_PAYLOAD = 0x40, CIP_HEADER_WITHOUT_EOH = 0x80, CIP_NO_HEADER = 0x100, + CIP_UNALIGHED_DBC = 0x200, }; /** @@ -91,12 +94,20 @@ enum amdtp_stream_direction { AMDTP_IN_STREAM }; +struct pkt_desc { + u32 cycle; + u32 syt; + unsigned int data_blocks; + unsigned int data_block_counter; + __be32 *ctx_payload; +}; + struct amdtp_stream; -typedef unsigned int (*amdtp_stream_process_data_blocks_t)( +typedef unsigned int (*amdtp_stream_process_ctx_payloads_t)( struct amdtp_stream *s, - __be32 *buffer, - unsigned int data_blocks, - unsigned int *syt); + const struct pkt_desc *desc, + unsigned int packets, + struct snd_pcm_substream *pcm); struct amdtp_stream { struct fw_unit *unit; enum cip_flags flags; @@ -107,6 +118,7 @@ struct amdtp_stream { struct fw_iso_context *context; struct iso_packets_buffer buffer; int packet_index; + struct pkt_desc *pkt_descs; int tag; union { struct { @@ -119,8 +131,6 @@ struct amdtp_stream { // Fixed interval of dbc between previos/current // packets. unsigned int dbc_interval; - // Indicate the value of dbc field in a first packet. - unsigned int first_dbc; } tx; struct { // To calculate CIP data blocks and tstamp. @@ -131,6 +141,7 @@ struct amdtp_stream { // To generate CIP header. unsigned int fdf; + int syt_override; } rx; } ctx_data; @@ -158,13 +169,18 @@ struct amdtp_stream { /* For backends to process data blocks. */ void *protocol; - amdtp_stream_process_data_blocks_t process_data_blocks; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; + + // For domain. + int channel; + int speed; + struct list_head list; }; int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, enum cip_flags flags, unsigned int fmt, - amdtp_stream_process_data_blocks_t process_data_blocks, + amdtp_stream_process_ctx_payloads_t process_ctx_payloads, unsigned int protocol_size); void amdtp_stream_destroy(struct amdtp_stream *s); @@ -172,9 +188,7 @@ int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate, unsigned int data_block_quadlets); unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s); -int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed); void amdtp_stream_update(struct amdtp_stream *s); -void amdtp_stream_stop(struct amdtp_stream *s); int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); @@ -256,4 +270,17 @@ static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s, msecs_to_jiffies(timeout)) > 0; } +struct amdtp_domain { + struct list_head streams; +}; + +int amdtp_domain_init(struct amdtp_domain *d); +void amdtp_domain_destroy(struct amdtp_domain *d); + +int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s, + int channel, int speed); + +int amdtp_domain_start(struct amdtp_domain *d); +void amdtp_domain_stop(struct amdtp_domain *d); + #endif diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 9e0b689fe34a..356d6ba60959 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -115,6 +115,8 @@ struct snd_bebob { /* For BeBoB version quirk. */ unsigned int version; + + struct amdtp_domain domain; }; static inline int diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 334dc7c96e1d..73fee991bd75 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -445,10 +445,9 @@ start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) goto end; } - /* start amdtp stream */ - err = amdtp_stream_start(stream, - conn->resources.channel, - conn->speed); + // start amdtp stream. + err = amdtp_domain_add_stream(&bebob->domain, stream, + conn->resources.channel, conn->speed); end: return err; } @@ -523,7 +522,13 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) return err; } - return 0; + 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, @@ -566,9 +571,7 @@ int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate) if (rate == 0) rate = curr_rate; if (curr_rate != rate) { - amdtp_stream_stop(&bebob->tx_stream); - amdtp_stream_stop(&bebob->rx_stream); - + amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); cmp_connection_release(&bebob->out_conn); @@ -620,9 +623,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) // packet queueing error or detecting discontinuity if (amdtp_streaming_error(&bebob->rx_stream) || amdtp_streaming_error(&bebob->tx_stream)) { - amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_stop(&bebob->tx_stream); - + amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); } @@ -640,11 +641,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) return err; err = start_stream(bebob, &bebob->rx_stream); - if (err < 0) { - dev_err(&bebob->unit->device, - "fail to run AMDTP master stream:%d\n", err); + if (err < 0) + goto error; + + err = start_stream(bebob, &bebob->tx_stream); + if (err < 0) + goto error; + + err = amdtp_domain_start(&bebob->domain); + if (err < 0) goto error; - } // NOTE: // The firmware customized by M-Audio uses these commands to @@ -660,21 +666,8 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) } if (!amdtp_stream_wait_callback(&bebob->rx_stream, - CALLBACK_TIMEOUT)) { - err = -ETIMEDOUT; - goto error; - } - } - - if (!amdtp_stream_running(&bebob->tx_stream)) { - err = start_stream(bebob, &bebob->tx_stream); - if (err < 0) { - dev_err(&bebob->unit->device, - "fail to run AMDTP slave stream:%d\n", err); - goto error; - } - - if (!amdtp_stream_wait_callback(&bebob->tx_stream, + CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(&bebob->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -683,8 +676,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) return 0; error: - amdtp_stream_stop(&bebob->tx_stream); - amdtp_stream_stop(&bebob->rx_stream); + amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); return err; } @@ -692,9 +684,7 @@ error: void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { if (bebob->substreams_counter == 0) { - amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_stop(&bebob->tx_stream); - + amdtp_domain_stop(&bebob->domain); break_both_connections(bebob); cmp_connection_release(&bebob->out_conn); @@ -708,6 +698,8 @@ void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) */ 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); } diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index 218292bdace6..f5b325263b67 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { static const unsigned int alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { - {10, 10, 8}, /* Tx0 = Analog + S/PDIF. */ + {10, 10, 4}, /* Tx0 = Analog + S/PDIF. */ {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */ }; diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index a9f0c77734c3..f6a8627ae5a2 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -154,14 +154,10 @@ static void stop_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, for (i = 0; i < params->count; i++) { reg = cpu_to_be32((u32)-1); if (dir == AMDTP_IN_STREAM) { - amdtp_stream_stop(&dice->tx_stream[i]); - snd_dice_transaction_write_tx(dice, params->size * i + TX_ISOCHRONOUS, ®, sizeof(reg)); } else { - amdtp_stream_stop(&dice->rx_stream[i]); - snd_dice_transaction_write_rx(dice, params->size * i + RX_ISOCHRONOUS, ®, sizeof(reg)); @@ -297,10 +293,11 @@ int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate) if (dice->substreams_counter == 0 || curr_rate != rate) { struct reg_params tx_params, rx_params; + amdtp_domain_stop(&dice->domain); + err = get_register_params(dice, &tx_params, &rx_params); if (err < 0) return err; - finish_session(dice, &tx_params, &rx_params); release_resources(dice); @@ -377,7 +374,8 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, return err; } - err = amdtp_stream_start(stream, resources->channel, max_speed); + err = amdtp_domain_add_stream(&dice->domain, stream, + resources->channel, max_speed); if (err < 0) return err; } @@ -410,6 +408,7 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice) for (i = 0; i < MAX_STREAMS; ++i) { if (amdtp_streaming_error(&dice->tx_stream[i]) || amdtp_streaming_error(&dice->rx_stream[i])) { + amdtp_domain_stop(&dice->domain); finish_session(dice, &tx_params, &rx_params); break; } @@ -456,6 +455,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice) goto error; } + err = amdtp_domain_start(&dice->domain); + if (err < 0) + goto error; + for (i = 0; i < MAX_STREAMS; i++) { if ((i < tx_params.count && !amdtp_stream_wait_callback(&dice->tx_stream[i], @@ -471,6 +474,7 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice) return 0; error: + amdtp_domain_stop(&dice->domain); finish_session(dice, &tx_params, &rx_params); return err; } @@ -485,8 +489,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice) struct reg_params tx_params, rx_params; if (dice->substreams_counter == 0) { - if (get_register_params(dice, &tx_params, &rx_params) >= 0) + if (get_register_params(dice, &tx_params, &rx_params) >= 0) { + amdtp_domain_stop(&dice->domain); finish_session(dice, &tx_params, &rx_params); + } release_resources(dice); } @@ -564,7 +570,15 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice) destroy_stream(dice, AMDTP_OUT_STREAM, i); for (i = 0; i < MAX_STREAMS; i++) destroy_stream(dice, AMDTP_IN_STREAM, i); - break; + goto end; + } + } + + err = amdtp_domain_init(&dice->domain); + if (err < 0) { + for (i = 0; i < MAX_STREAMS; ++i) { + destroy_stream(dice, AMDTP_OUT_STREAM, i); + destroy_stream(dice, AMDTP_IN_STREAM, i); } } end: @@ -579,6 +593,8 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice) destroy_stream(dice, AMDTP_IN_STREAM, i); destroy_stream(dice, AMDTP_OUT_STREAM, i); } + + amdtp_domain_destroy(&dice->domain); } void snd_dice_stream_update_duplex(struct snd_dice *dice) @@ -596,6 +612,8 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice) dice->global_enabled = false; if (get_register_params(dice, &tx_params, &rx_params) == 0) { + amdtp_domain_stop(&dice->domain); + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index c6304e5e9fc4..fa6d74303f54 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -112,6 +112,8 @@ struct snd_dice { bool global_enabled; struct completion clock_accepted; unsigned int substreams_counter; + + struct amdtp_domain domain; }; enum snd_dice_addr_type { diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index 45ff73d16074..d613642a2ce3 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -143,17 +143,23 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate, } static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_dot *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; const u32 *src; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; buffer++; for (i = 0; i < frames; ++i) { @@ -169,17 +175,23 @@ static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, } static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_dot *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; u32 *dst; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; buffer++; for (i = 0; i < frames; ++i) { @@ -234,7 +246,7 @@ static inline void midi_use_bytes(struct amdtp_stream *s, } static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, - unsigned int data_blocks) + unsigned int data_blocks, unsigned int data_block_counter) { struct amdtp_dot *p = s->protocol; unsigned int f, port; @@ -242,7 +254,7 @@ static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer, u8 *b; for (f = 0; f < data_blocks; f++) { - port = (s->data_block_counter + f) % 8; + port = (data_block_counter + f) % 8; b = (u8 *)&buffer[0]; len = 0; @@ -329,45 +341,53 @@ void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, WRITE_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) +static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm; - unsigned int pcm_frames; + unsigned int pcm_frames = 0; + int i; - pcm = READ_ONCE(s->pcm); - if (pcm) { - read_pcm_s32(s, pcm, buffer, data_blocks); - pcm_frames = data_blocks; - } else { - pcm_frames = 0; - } + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - read_midi_messages(s, buffer, data_blocks); + if (pcm) { + read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } + + read_midi_messages(s, buf, 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) +static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm; - unsigned int pcm_frames; + unsigned int pcm_frames = 0; + int i; - pcm = READ_ONCE(s->pcm); - if (pcm) { - write_pcm_s32(s, pcm, buffer, data_blocks); - pcm_frames = data_blocks; - } else { - write_pcm_silence(s, buffer, data_blocks); - pcm_frames = 0; - } + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - write_midi_messages(s, buffer, data_blocks); + if (pcm) { + write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } else { + write_pcm_silence(s, buf, data_blocks); + } + + write_midi_messages(s, buf, data_blocks, + desc->data_block_counter); + } return pcm_frames; } @@ -375,20 +395,20 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, 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; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; enum cip_flags flags; - /* Use different mode between incoming/outgoing. */ + // Use different mode between incoming/outgoing. if (dir == AMDTP_IN_STREAM) { flags = CIP_NONBLOCKING; - process_data_blocks = process_tx_data_blocks; + process_ctx_payloads = process_ir_ctx_payloads; } else { flags = CIP_BLOCKING; - process_data_blocks = process_rx_data_blocks; + process_ctx_payloads = process_it_ctx_payloads; } return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM, - process_data_blocks, sizeof(struct amdtp_dot)); + process_ctx_payloads, sizeof(struct amdtp_dot)); } void amdtp_dot_reset(struct amdtp_stream *s) diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index 3e77dbd3ee22..d6a92460060f 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -126,9 +126,6 @@ static void finish_session(struct snd_dg00x *dg00x) { __be32 data; - amdtp_stream_stop(&dg00x->tx_stream); - amdtp_stream_stop(&dg00x->rx_stream); - data = cpu_to_be32(0x00000003); snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST, DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET, @@ -218,29 +215,59 @@ static int keep_resources(struct snd_dg00x *dg00x, struct amdtp_stream *stream, fw_parent_device(dg00x->unit)->max_speed); } -int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x) +static int init_stream(struct snd_dg00x *dg00x, struct amdtp_stream *s) { + struct fw_iso_resources *resources; + enum amdtp_stream_direction dir; int err; - /* For out-stream. */ - err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit); + if (s == &dg00x->tx_stream) { + resources = &dg00x->tx_resources; + dir = AMDTP_IN_STREAM; + } else { + resources = &dg00x->rx_resources; + dir = AMDTP_OUT_STREAM; + } + + err = fw_iso_resources_init(resources, dg00x->unit); if (err < 0) - goto error; - err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM); + return err; + + err = amdtp_dot_init(s, dg00x->unit, dir); if (err < 0) - goto error; + fw_iso_resources_destroy(resources); + + return err; +} + +static void destroy_stream(struct snd_dg00x *dg00x, struct amdtp_stream *s) +{ + amdtp_stream_destroy(s); + + if (s == &dg00x->tx_stream) + fw_iso_resources_destroy(&dg00x->tx_resources); + else + fw_iso_resources_destroy(&dg00x->rx_resources); +} + +int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x) +{ + int err; - /* For in-stream. */ - err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit); + err = init_stream(dg00x, &dg00x->rx_stream); if (err < 0) - goto error; - err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM); + return err; + + err = init_stream(dg00x, &dg00x->tx_stream); if (err < 0) - goto error; + destroy_stream(dg00x, &dg00x->rx_stream); + + err = amdtp_domain_init(&dg00x->domain); + if (err < 0) { + destroy_stream(dg00x, &dg00x->rx_stream); + destroy_stream(dg00x, &dg00x->tx_stream); + } - return 0; -error: - snd_dg00x_stream_destroy_duplex(dg00x); return err; } @@ -250,11 +277,10 @@ error: */ void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x) { - amdtp_stream_destroy(&dg00x->rx_stream); - fw_iso_resources_destroy(&dg00x->rx_resources); + amdtp_domain_destroy(&dg00x->domain); - amdtp_stream_destroy(&dg00x->tx_stream); - fw_iso_resources_destroy(&dg00x->tx_resources); + destroy_stream(dg00x, &dg00x->rx_stream); + destroy_stream(dg00x, &dg00x->tx_stream); } int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate) @@ -269,6 +295,8 @@ int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate) rate = curr_rate; if (dg00x->substreams_counter == 0 || curr_rate != rate) { + amdtp_domain_stop(&dg00x->domain); + finish_session(dg00x); fw_iso_resources_free(&dg00x->tx_resources); @@ -301,8 +329,10 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x) return 0; if (amdtp_streaming_error(&dg00x->tx_stream) || - amdtp_streaming_error(&dg00x->rx_stream)) + amdtp_streaming_error(&dg00x->rx_stream)) { + amdtp_domain_stop(&dg00x->domain); finish_session(dg00x); + } if (generation != fw_parent_device(dg00x->unit)->card->generation) { err = fw_iso_resources_update(&dg00x->tx_resources); @@ -319,36 +349,30 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x) * which source of clock is used. */ if (!amdtp_stream_running(&dg00x->rx_stream)) { + int spd = fw_parent_device(dg00x->unit)->max_speed; + err = begin_session(dg00x); if (err < 0) goto error; - err = amdtp_stream_start(&dg00x->rx_stream, - dg00x->rx_resources.channel, - fw_parent_device(dg00x->unit)->max_speed); + err = amdtp_domain_add_stream(&dg00x->domain, &dg00x->rx_stream, + dg00x->rx_resources.channel, spd); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&dg00x->rx_stream, - CALLBACK_TIMEOUT)) { - err = -ETIMEDOUT; + err = amdtp_domain_add_stream(&dg00x->domain, &dg00x->tx_stream, + dg00x->tx_resources.channel, spd); + if (err < 0) goto error; - } - } - /* - * The value of SYT field in transmitted packets is always 0x0000. Thus, - * duplex streams with timestamp synchronization cannot be built. - */ - if (!amdtp_stream_running(&dg00x->tx_stream)) { - err = amdtp_stream_start(&dg00x->tx_stream, - dg00x->tx_resources.channel, - fw_parent_device(dg00x->unit)->max_speed); + err = amdtp_domain_start(&dg00x->domain); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&dg00x->tx_stream, - CALLBACK_TIMEOUT)) { + if (!amdtp_stream_wait_callback(&dg00x->rx_stream, + CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(&dg00x->tx_stream, + CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } @@ -356,6 +380,7 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x) return 0; error: + amdtp_domain_stop(&dg00x->domain); finish_session(dg00x); return err; @@ -364,6 +389,7 @@ error: void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x) { if (dg00x->substreams_counter == 0) { + amdtp_domain_stop(&dg00x->domain); finish_session(dg00x); fw_iso_resources_free(&dg00x->tx_resources); diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 0994d191ccda..8041c65f2736 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -59,6 +59,8 @@ struct snd_dg00x { /* Console models have additional MIDI ports for control surface. */ bool is_console; + + struct amdtp_domain domain; }; #define DG00X_ADDR_BASE 0xffffe0000000ull diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c index 2938489740b4..119c0076b17a 100644 --- a/sound/firewire/fireface/amdtp-ff.c +++ b/sound/firewire/fireface/amdtp-ff.c @@ -27,19 +27,24 @@ int amdtp_ff_set_parameters(struct amdtp_stream *s, unsigned int rate, return amdtp_stream_set_parameters(s, rate, data_channels); } -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __le32 *buffer, unsigned int frames) +static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __le32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_ff *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; const u32 *src; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { @@ -52,19 +57,24 @@ static void write_pcm_s32(struct amdtp_stream *s, } } -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __le32 *buffer, unsigned int frames) +static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __le32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_ff *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; u32 *dst; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { @@ -102,38 +112,47 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s, return amdtp_stream_add_pcm_hw_constraints(s, runtime); } -static unsigned int process_rx_data_blocks(struct amdtp_stream *s, - __be32 *buffer, - unsigned int data_blocks, - unsigned int *syt) +static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); - unsigned int pcm_frames; - - if (pcm) { - write_pcm_s32(s, pcm, (__le32 *)buffer, data_blocks); - pcm_frames = data_blocks; - } else { - write_pcm_silence(s, (__le32 *)buffer, data_blocks); - pcm_frames = 0; + unsigned int pcm_frames = 0; + int i; + + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __le32 *buf = (__le32 *)desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; + + if (pcm) { + write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } else { + write_pcm_silence(s, buf, data_blocks); + } } return pcm_frames; } -static unsigned int process_tx_data_blocks(struct amdtp_stream *s, - __be32 *buffer, - unsigned int data_blocks, - unsigned int *syt) +static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); - unsigned int pcm_frames; - - if (pcm) { - read_pcm_s32(s, pcm, (__le32 *)buffer, data_blocks); - pcm_frames = data_blocks; - } else { - pcm_frames = 0; + unsigned int pcm_frames = 0; + int i; + + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __le32 *buf = (__le32 *)desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; + + if (pcm) { + read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } } return pcm_frames; @@ -142,13 +161,13 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, int amdtp_ff_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir) { - amdtp_stream_process_data_blocks_t process_data_blocks; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; if (dir == AMDTP_IN_STREAM) - process_data_blocks = process_tx_data_blocks; + process_ctx_payloads = process_ir_ctx_payloads; else - process_data_blocks = process_rx_data_blocks; + process_ctx_payloads = process_it_ctx_payloads; return amdtp_stream_init(s, unit, dir, CIP_NO_HEADER, 0, - process_data_blocks, sizeof(struct amdtp_ff)); + process_ctx_payloads, sizeof(struct amdtp_ff)); } diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index 4208b8004d1a..e8e6f9fd6433 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -32,61 +32,65 @@ int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc, static inline void finish_session(struct snd_ff *ff) { - amdtp_stream_stop(&ff->tx_stream); - amdtp_stream_stop(&ff->rx_stream); - ff->spec->protocol->finish_session(ff); ff->spec->protocol->switch_fetching_mode(ff, false); } -static int init_stream(struct snd_ff *ff, enum amdtp_stream_direction dir) +static int init_stream(struct snd_ff *ff, struct amdtp_stream *s) { - int err; struct fw_iso_resources *resources; - struct amdtp_stream *stream; + enum amdtp_stream_direction dir; + int err; - if (dir == AMDTP_IN_STREAM) { + if (s == &ff->tx_stream) { resources = &ff->tx_resources; - stream = &ff->tx_stream; + dir = AMDTP_IN_STREAM; } else { resources = &ff->rx_resources; - stream = &ff->rx_stream; + dir = AMDTP_OUT_STREAM; } err = fw_iso_resources_init(resources, ff->unit); if (err < 0) return err; - err = amdtp_ff_init(stream, ff->unit, dir); + err = amdtp_ff_init(s, ff->unit, dir); if (err < 0) fw_iso_resources_destroy(resources); return err; } -static void destroy_stream(struct snd_ff *ff, enum amdtp_stream_direction dir) +static void destroy_stream(struct snd_ff *ff, struct amdtp_stream *s) { - if (dir == AMDTP_IN_STREAM) { - amdtp_stream_destroy(&ff->tx_stream); + amdtp_stream_destroy(s); + + if (s == &ff->tx_stream) fw_iso_resources_destroy(&ff->tx_resources); - } else { - amdtp_stream_destroy(&ff->rx_stream); + else fw_iso_resources_destroy(&ff->rx_resources); - } } int snd_ff_stream_init_duplex(struct snd_ff *ff) { int err; - err = init_stream(ff, AMDTP_OUT_STREAM); + err = init_stream(ff, &ff->rx_stream); if (err < 0) - goto end; + return err; + + err = init_stream(ff, &ff->tx_stream); + if (err < 0) { + destroy_stream(ff, &ff->rx_stream); + return err; + } + + err = amdtp_domain_init(&ff->domain); + if (err < 0) { + destroy_stream(ff, &ff->rx_stream); + destroy_stream(ff, &ff->tx_stream); + } - err = init_stream(ff, AMDTP_IN_STREAM); - if (err < 0) - destroy_stream(ff, AMDTP_OUT_STREAM); -end: return err; } @@ -96,8 +100,10 @@ end: */ void snd_ff_stream_destroy_duplex(struct snd_ff *ff) { - destroy_stream(ff, AMDTP_IN_STREAM); - destroy_stream(ff, AMDTP_OUT_STREAM); + amdtp_domain_destroy(&ff->domain); + + destroy_stream(ff, &ff->rx_stream); + destroy_stream(ff, &ff->tx_stream); } int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate) @@ -114,6 +120,7 @@ int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate) enum snd_ff_stream_mode mode; int i; + amdtp_domain_stop(&ff->domain); finish_session(ff); fw_iso_resources_free(&ff->tx_resources); @@ -156,51 +163,52 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) return 0; if (amdtp_streaming_error(&ff->tx_stream) || - amdtp_streaming_error(&ff->rx_stream)) + amdtp_streaming_error(&ff->rx_stream)) { + amdtp_domain_stop(&ff->domain); finish_session(ff); + } /* * Regardless of current source of clock signal, drivers transfer some * packets. Then, the device transfers packets. */ if (!amdtp_stream_running(&ff->rx_stream)) { + int spd = fw_parent_device(ff->unit)->max_speed; + err = ff->spec->protocol->begin_session(ff, rate); if (err < 0) goto error; - err = amdtp_stream_start(&ff->rx_stream, - ff->rx_resources.channel, - fw_parent_device(ff->unit)->max_speed); + err = amdtp_domain_add_stream(&ff->domain, &ff->rx_stream, + ff->rx_resources.channel, spd); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&ff->rx_stream, - CALLBACK_TIMEOUT_MS)) { - err = -ETIMEDOUT; - goto error; - } - - err = ff->spec->protocol->switch_fetching_mode(ff, true); + err = amdtp_domain_add_stream(&ff->domain, &ff->tx_stream, + ff->tx_resources.channel, spd); if (err < 0) goto error; - } - if (!amdtp_stream_running(&ff->tx_stream)) { - err = amdtp_stream_start(&ff->tx_stream, - ff->tx_resources.channel, - fw_parent_device(ff->unit)->max_speed); + err = amdtp_domain_start(&ff->domain); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&ff->tx_stream, + if (!amdtp_stream_wait_callback(&ff->rx_stream, + CALLBACK_TIMEOUT_MS) || + !amdtp_stream_wait_callback(&ff->tx_stream, CALLBACK_TIMEOUT_MS)) { err = -ETIMEDOUT; goto error; } + + err = ff->spec->protocol->switch_fetching_mode(ff, true); + if (err < 0) + goto error; } return 0; error: + amdtp_domain_stop(&ff->domain); finish_session(ff); return err; @@ -209,6 +217,7 @@ error: void snd_ff_stream_stop_duplex(struct snd_ff *ff) { if (ff->substreams_counter == 0) { + amdtp_domain_stop(&ff->domain); finish_session(ff); fw_iso_resources_free(&ff->tx_resources); @@ -218,12 +227,11 @@ void snd_ff_stream_stop_duplex(struct snd_ff *ff) void snd_ff_stream_update_duplex(struct snd_ff *ff) { + amdtp_domain_stop(&ff->domain); + // The device discontinue to transfer packets. amdtp_stream_pcm_abort(&ff->tx_stream); - amdtp_stream_stop(&ff->tx_stream); - amdtp_stream_pcm_abort(&ff->rx_stream); - amdtp_stream_stop(&ff->rx_stream); } void snd_ff_stream_lock_changed(struct snd_ff *ff) diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 36dd0c75b9f7..b4c22ca6079e 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -91,6 +91,8 @@ struct snd_ff { int dev_lock_count; bool dev_lock_changed; wait_queue_head_t hwdep_wait; + + struct amdtp_domain domain; }; enum snd_ff_clock_src { diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 31efd4b53b4f..4cda297f8438 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -107,6 +107,8 @@ struct snd_efw { u8 *resp_buf; u8 *pull_ptr; u8 *push_ptr; + + struct amdtp_domain domain; }; int snd_efw_transaction_cmd(struct fw_unit *unit, diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index e659a0b89ba5..f2de304d2f26 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -8,8 +8,7 @@ #define CALLBACK_TIMEOUT 100 -static int -init_stream(struct snd_efw *efw, struct amdtp_stream *stream) +static int init_stream(struct snd_efw *efw, struct amdtp_stream *stream) { struct cmp_connection *conn; enum cmp_direction c_dir; @@ -28,26 +27,38 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream) err = cmp_connection_init(conn, efw->unit, c_dir, 0); if (err < 0) - goto end; + return err; err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING); if (err < 0) { amdtp_stream_destroy(stream); cmp_connection_destroy(conn); + return err; } -end: - return err; -} -static void -stop_stream(struct snd_efw *efw, struct amdtp_stream *stream) -{ - amdtp_stream_stop(stream); + if (stream == &efw->tx_stream) { + // Fireworks transmits NODATA packets with TAG0. + efw->tx_stream.flags |= CIP_EMPTY_WITH_TAG0; + // Fireworks has its own meaning for dbc. + efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; + // Fireworks reset dbc at bus reset. + efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + // But Recent firmwares starts packets with non-zero dbc. + // Driver version 5.7.6 installs firmware version 5.7.3. + if (efw->is_fireworks3 && + (efw->firmware_version == 0x5070000 || + efw->firmware_version == 0x5070300 || + efw->firmware_version == 0x5080000)) + efw->tx_stream.flags |= CIP_UNALIGHED_DBC; + // AudioFire9 always reports wrong dbs. + if (efw->is_af9) + efw->tx_stream.flags |= CIP_WRONG_DBS; + // Firmware version 5.5 reports fixed interval for dbc. + if (efw->firmware_version == 0x5050000) + efw->tx_stream.ctx_data.tx.dbc_interval = 8; + } - if (stream == &efw->tx_stream) - cmp_connection_break(&efw->out_conn); - else - cmp_connection_break(&efw->in_conn); + return err; } static int start_stream(struct snd_efw *efw, struct amdtp_stream *stream, @@ -67,38 +78,26 @@ static int start_stream(struct snd_efw *efw, struct amdtp_stream *stream, return err; // Start amdtp stream. - err = amdtp_stream_start(stream, conn->resources.channel, conn->speed); + err = amdtp_domain_add_stream(&efw->domain, stream, + conn->resources.channel, conn->speed); if (err < 0) { cmp_connection_break(conn); return err; } - // Wait first callback. - if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(stream); - cmp_connection_break(conn); - return -ETIMEDOUT; - } - return 0; } -/* - * This function should be called before starting the stream or after stopping - * the streams. - */ -static void -destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) +// This function should be called before starting the stream or after stopping +// the streams. +static void destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) { - struct cmp_connection *conn; + amdtp_stream_destroy(stream); if (stream == &efw->tx_stream) - conn = &efw->out_conn; + cmp_connection_destroy(&efw->out_conn); else - conn = &efw->in_conn; - - amdtp_stream_destroy(stream); - cmp_connection_destroy(conn); + cmp_connection_destroy(&efw->in_conn); } static int @@ -131,42 +130,28 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) err = init_stream(efw, &efw->tx_stream); if (err < 0) - goto end; - /* Fireworks transmits NODATA packets with TAG0. */ - efw->tx_stream.flags |= CIP_EMPTY_WITH_TAG0; - /* Fireworks has its own meaning for dbc. */ - efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; - /* Fireworks reset dbc at bus reset. */ - efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; - /* - * But Recent firmwares starts packets with non-zero dbc. - * Driver version 5.7.6 installs firmware version 5.7.3. - */ - if (efw->is_fireworks3 && - (efw->firmware_version == 0x5070000 || - efw->firmware_version == 0x5070300 || - efw->firmware_version == 0x5080000)) - efw->tx_stream.ctx_data.tx.first_dbc = 0x02; - /* AudioFire9 always reports wrong dbs. */ - if (efw->is_af9) - efw->tx_stream.flags |= CIP_WRONG_DBS; - /* Firmware version 5.5 reports fixed interval for dbc. */ - if (efw->firmware_version == 0x5050000) - efw->tx_stream.ctx_data.tx.dbc_interval = 8; + return err; err = init_stream(efw, &efw->rx_stream); if (err < 0) { destroy_stream(efw, &efw->tx_stream); - goto end; + return err; + } + + err = amdtp_domain_init(&efw->domain); + if (err < 0) { + destroy_stream(efw, &efw->tx_stream); + destroy_stream(efw, &efw->rx_stream); + return err; } - /* set IEC61883 compliant mode (actually not fully compliant...) */ + // set IEC61883 compliant mode (actually not fully compliant...). err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883); if (err < 0) { destroy_stream(efw, &efw->tx_stream); destroy_stream(efw, &efw->rx_stream); } -end: + return err; } @@ -214,8 +199,10 @@ int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - stop_stream(efw, &efw->tx_stream); - stop_stream(efw, &efw->rx_stream); + amdtp_domain_stop(&efw->domain); + + cmp_connection_break(&efw->out_conn); + cmp_connection_break(&efw->in_conn); cmp_connection_release(&efw->out_conn); cmp_connection_release(&efw->in_conn); @@ -255,47 +242,57 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw) if (efw->substreams_counter == 0) return -EIO; - err = snd_efw_command_get_sampling_rate(efw, &rate); - if (err < 0) - return err; - if (amdtp_streaming_error(&efw->rx_stream) || amdtp_streaming_error(&efw->tx_stream)) { - stop_stream(efw, &efw->rx_stream); - stop_stream(efw, &efw->tx_stream); + amdtp_domain_stop(&efw->domain); + cmp_connection_break(&efw->out_conn); + cmp_connection_break(&efw->in_conn); } - /* master should be always running */ + err = snd_efw_command_get_sampling_rate(efw, &rate); + if (err < 0) + return err; + if (!amdtp_stream_running(&efw->rx_stream)) { err = start_stream(efw, &efw->rx_stream, rate); - if (err < 0) { - dev_err(&efw->unit->device, - "fail to start AMDTP master stream:%d\n", err); + if (err < 0) goto error; - } - } - if (!amdtp_stream_running(&efw->tx_stream)) { err = start_stream(efw, &efw->tx_stream, rate); - if (err < 0) { - dev_err(&efw->unit->device, - "fail to start AMDTP slave stream:%d\n", err); + if (err < 0) + goto error; + + err = amdtp_domain_start(&efw->domain); + if (err < 0) + goto error; + + // Wait first callback. + if (!amdtp_stream_wait_callback(&efw->rx_stream, + CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(&efw->tx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; goto error; } } return 0; error: - stop_stream(efw, &efw->rx_stream); - stop_stream(efw, &efw->tx_stream); + amdtp_domain_stop(&efw->domain); + + cmp_connection_break(&efw->out_conn); + cmp_connection_break(&efw->in_conn); + return err; } void snd_efw_stream_stop_duplex(struct snd_efw *efw) { if (efw->substreams_counter == 0) { - stop_stream(efw, &efw->tx_stream); - stop_stream(efw, &efw->rx_stream); + amdtp_domain_stop(&efw->domain); + + cmp_connection_break(&efw->out_conn); + cmp_connection_break(&efw->in_conn); cmp_connection_release(&efw->out_conn); cmp_connection_release(&efw->in_conn); @@ -304,18 +301,19 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw) void snd_efw_stream_update_duplex(struct snd_efw *efw) { - if (cmp_connection_update(&efw->out_conn) < 0 || - cmp_connection_update(&efw->in_conn) < 0) { - stop_stream(efw, &efw->rx_stream); - stop_stream(efw, &efw->tx_stream); - } else { - amdtp_stream_update(&efw->rx_stream); - amdtp_stream_update(&efw->tx_stream); - } + amdtp_domain_stop(&efw->domain); + + cmp_connection_break(&efw->out_conn); + cmp_connection_break(&efw->in_conn); + + amdtp_stream_pcm_abort(&efw->rx_stream); + amdtp_stream_pcm_abort(&efw->tx_stream); } void snd_efw_stream_destroy_duplex(struct snd_efw *efw) { + amdtp_domain_destroy(&efw->domain); + destroy_stream(efw, &efw->rx_stream); destroy_stream(efw, &efw->tx_stream); } diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 7973dedd31ef..0fd36e469ad0 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -117,19 +117,25 @@ int amdtp_motu_set_parameters(struct amdtp_stream *s, unsigned int rate, return 0; } -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime, - __be32 *buffer, unsigned int data_blocks) +static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int data_blocks, + unsigned int pcm_frames) { struct amdtp_motu *p = s->protocol; - unsigned int channels, remaining_frames, i, c; + unsigned int channels = p->pcm_chunks; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int pcm_buffer_pointer; + int remaining_frames; u8 *byte; u32 *dst; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - channels = p->pcm_chunks; dst = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < data_blocks; ++i) { byte = (u8 *)buffer + p->pcm_byte_offset; @@ -147,19 +153,25 @@ static void read_pcm_s32(struct amdtp_stream *s, } } -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime, - __be32 *buffer, unsigned int data_blocks) +static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int data_blocks, + unsigned int pcm_frames) { struct amdtp_motu *p = s->protocol; - unsigned int channels, remaining_frames, i, c; + unsigned int channels = p->pcm_chunks; + struct snd_pcm_runtime *runtime = pcm->runtime; + unsigned int pcm_buffer_pointer; + int remaining_frames; u8 *byte; const u32 *src; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - channels = p->pcm_chunks; src = (void *)runtime->dma_area + - frames_to_bytes(runtime, s->pcm_buffer_pointer); - remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < data_blocks; ++i) { byte = (u8 *)buffer + p->pcm_byte_offset; @@ -298,24 +310,52 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, } } -static unsigned int process_tx_data_blocks(struct amdtp_stream *s, - __be32 *buffer, unsigned int data_blocks, - unsigned int *syt) +static void probe_tracepoints_events(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets) +{ + int i; + + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; + + trace_data_block_sph(s, data_blocks, buf); + trace_data_block_message(s, data_blocks, buf); + } +} + +static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { struct amdtp_motu *p = s->protocol; - struct snd_pcm_substream *pcm; + unsigned int pcm_frames = 0; + int i; + + // For data block processing. + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - trace_data_block_sph(s, data_blocks, buffer); - trace_data_block_message(s, data_blocks, buffer); + if (pcm) { + read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } - if (p->midi_ports) - read_midi_messages(s, buffer, data_blocks); + if (p->midi_ports) + read_midi_messages(s, buf, data_blocks); + } - pcm = READ_ONCE(s->pcm); - if (data_blocks > 0 && pcm) - read_pcm_s32(s, pcm->runtime, buffer, data_blocks); + // For tracepoints. + if (trace_data_block_sph_enabled() || + trace_data_block_message_enabled()) + probe_tracepoints_events(s, descs, packets); - return data_blocks; + return pcm_frames; } static inline void compute_next_elapse_from_start(struct amdtp_motu *p) @@ -360,46 +400,55 @@ static void write_sph(struct amdtp_stream *s, __be32 *buffer, } } -static unsigned int process_rx_data_blocks(struct amdtp_stream *s, - __be32 *buffer, unsigned int data_blocks, - unsigned int *syt) +static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct amdtp_motu *p = (struct amdtp_motu *)s->protocol; - struct snd_pcm_substream *pcm; + struct amdtp_motu *p = s->protocol; + unsigned int pcm_frames = 0; + int i; - /* Not used. */ - *syt = 0xffff; + // For data block processing. + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - /* TODO: how to interact control messages between userspace? */ + if (pcm) { + write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } else { + write_pcm_silence(s, buf, data_blocks); + } - if (p->midi_ports) - write_midi_messages(s, buffer, data_blocks); + if (p->midi_ports) + write_midi_messages(s, buf, data_blocks); - pcm = READ_ONCE(s->pcm); - if (pcm) - write_pcm_s32(s, pcm->runtime, buffer, data_blocks); - else - write_pcm_silence(s, buffer, data_blocks); + // TODO: how to interact control messages between userspace? - write_sph(s, buffer, data_blocks); + write_sph(s, buf, data_blocks); + } - trace_data_block_sph(s, data_blocks, buffer); - trace_data_block_message(s, data_blocks, buffer); + // For tracepoints. + if (trace_data_block_sph_enabled() || + trace_data_block_message_enabled()) + probe_tracepoints_events(s, descs, packets); - return data_blocks; + return pcm_frames; } int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, const struct snd_motu_protocol *const protocol) { - amdtp_stream_process_data_blocks_t process_data_blocks; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; int fmt = CIP_FMT_MOTU; int flags = CIP_BLOCKING; int err; if (dir == AMDTP_IN_STREAM) { - process_data_blocks = process_tx_data_blocks; + process_ctx_payloads = process_ir_ctx_payloads; /* * Units of version 3 transmits packets with invalid CIP header @@ -418,17 +467,23 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, CIP_SKIP_DBC_ZERO_CHECK; } } else { - process_data_blocks = process_rx_data_blocks; + process_ctx_payloads = process_it_ctx_payloads; flags |= CIP_DBC_IS_END_EVENT; } - err = amdtp_stream_init(s, unit, dir, flags, fmt, process_data_blocks, + err = amdtp_stream_init(s, unit, dir, flags, fmt, process_ctx_payloads, sizeof(struct amdtp_motu)); if (err < 0) return err; s->sph = 1; - s->ctx_data.rx.fdf = MOTU_FDF_AM824; + + if (dir == AMDTP_OUT_STREAM) { + // Use fixed value for FDF field. + s->ctx_data.rx.fdf = MOTU_FDF_AM824; + // Not used. + s->ctx_data.rx.syt_override = 0xffff; + } return 0; } diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 2bbb335e8de1..813e38e6a86e 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -92,9 +92,6 @@ static void finish_session(struct snd_motu *motu) if (err < 0) return; - amdtp_stream_stop(&motu->tx_stream); - amdtp_stream_stop(&motu->rx_stream); - err = snd_motu_transaction_read(motu, ISOC_COMM_CONTROL_OFFSET, ®, sizeof(reg)); if (err < 0) @@ -109,27 +106,6 @@ static void finish_session(struct snd_motu *motu) sizeof(reg)); } -static int start_isoc_ctx(struct snd_motu *motu, struct amdtp_stream *stream) -{ - struct fw_iso_resources *resources; - int err; - - if (stream == &motu->rx_stream) - resources = &motu->rx_resources; - else - resources = &motu->tx_resources; - - err = amdtp_stream_start(stream, resources->channel, - fw_parent_device(motu->unit)->max_speed); - if (err < 0) - return err; - - if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) - return -ETIMEDOUT; - - return 0; -} - int snd_motu_stream_cache_packet_formats(struct snd_motu *motu) { int err; @@ -169,6 +145,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate) rate = curr_rate; if (motu->substreams_counter == 0 || curr_rate != rate) { + amdtp_domain_stop(&motu->domain); finish_session(motu); fw_iso_resources_free(&motu->tx_resources); @@ -234,8 +211,10 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu) return 0; if (amdtp_streaming_error(&motu->rx_stream) || - amdtp_streaming_error(&motu->tx_stream)) + amdtp_streaming_error(&motu->tx_stream)) { + amdtp_domain_stop(&motu->domain); finish_session(motu); + } if (generation != fw_parent_device(motu->unit)->card->generation) { err = fw_iso_resources_update(&motu->rx_resources); @@ -248,6 +227,8 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu) } if (!amdtp_stream_running(&motu->rx_stream)) { + int spd = fw_parent_device(motu->unit)->max_speed; + err = ensure_packet_formats(motu); if (err < 0) return err; @@ -259,26 +240,32 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu) goto stop_streams; } - err = start_isoc_ctx(motu, &motu->rx_stream); - if (err < 0) { - dev_err(&motu->unit->device, - "fail to start IT context: %d\n", err); + err = amdtp_domain_add_stream(&motu->domain, &motu->tx_stream, + motu->tx_resources.channel, spd); + if (err < 0) goto stop_streams; - } - err = motu->spec->protocol->switch_fetching_mode(motu, true); - if (err < 0) { - dev_err(&motu->unit->device, - "fail to enable frame fetching: %d\n", err); + err = amdtp_domain_add_stream(&motu->domain, &motu->rx_stream, + motu->rx_resources.channel, spd); + if (err < 0) + goto stop_streams; + + err = amdtp_domain_start(&motu->domain); + if (err < 0) + goto stop_streams; + + if (!amdtp_stream_wait_callback(&motu->tx_stream, + CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(&motu->rx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; goto stop_streams; } - } - if (!amdtp_stream_running(&motu->tx_stream)) { - err = start_isoc_ctx(motu, &motu->tx_stream); + err = motu->spec->protocol->switch_fetching_mode(motu, true); if (err < 0) { dev_err(&motu->unit->device, - "fail to start IR context: %d", err); + "fail to enable frame fetching: %d\n", err); goto stop_streams; } } @@ -286,6 +273,7 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu) return 0; stop_streams: + amdtp_domain_stop(&motu->domain); finish_session(motu); return err; } @@ -293,6 +281,7 @@ stop_streams: void snd_motu_stream_stop_duplex(struct snd_motu *motu) { if (motu->substreams_counter == 0) { + amdtp_domain_stop(&motu->domain); finish_session(motu); fw_iso_resources_free(&motu->tx_resources); @@ -300,74 +289,72 @@ void snd_motu_stream_stop_duplex(struct snd_motu *motu) } } -static int init_stream(struct snd_motu *motu, enum amdtp_stream_direction dir) +static int init_stream(struct snd_motu *motu, struct amdtp_stream *s) { - int err; - struct amdtp_stream *stream; struct fw_iso_resources *resources; + enum amdtp_stream_direction dir; + int err; - if (dir == AMDTP_IN_STREAM) { - stream = &motu->tx_stream; + if (s == &motu->tx_stream) { resources = &motu->tx_resources; + dir = AMDTP_IN_STREAM; } else { - stream = &motu->rx_stream; resources = &motu->rx_resources; + dir = AMDTP_OUT_STREAM; } err = fw_iso_resources_init(resources, motu->unit); if (err < 0) return err; - err = amdtp_motu_init(stream, motu->unit, dir, motu->spec->protocol); - if (err < 0) { - amdtp_stream_destroy(stream); + err = amdtp_motu_init(s, motu->unit, dir, motu->spec->protocol); + if (err < 0) fw_iso_resources_destroy(resources); - } return err; } -static void destroy_stream(struct snd_motu *motu, - enum amdtp_stream_direction dir) +static void destroy_stream(struct snd_motu *motu, struct amdtp_stream *s) { - struct amdtp_stream *stream; - struct fw_iso_resources *resources; - - if (dir == AMDTP_IN_STREAM) { - stream = &motu->tx_stream; - resources = &motu->tx_resources; - } else { - stream = &motu->rx_stream; - resources = &motu->rx_resources; - } + amdtp_stream_destroy(s); - amdtp_stream_destroy(stream); - fw_iso_resources_destroy(resources); + if (s == &motu->tx_stream) + fw_iso_resources_destroy(&motu->tx_resources); + else + fw_iso_resources_destroy(&motu->rx_resources); } int snd_motu_stream_init_duplex(struct snd_motu *motu) { int err; - err = init_stream(motu, AMDTP_IN_STREAM); + err = init_stream(motu, &motu->tx_stream); if (err < 0) return err; - err = init_stream(motu, AMDTP_OUT_STREAM); - if (err < 0) - destroy_stream(motu, AMDTP_IN_STREAM); + err = init_stream(motu, &motu->rx_stream); + if (err < 0) { + destroy_stream(motu, &motu->tx_stream); + return err; + } + + err = amdtp_domain_init(&motu->domain); + if (err < 0) { + destroy_stream(motu, &motu->tx_stream); + destroy_stream(motu, &motu->rx_stream); + } return err; } -/* - * This function should be called before starting streams or after stopping - * streams. - */ +// This function should be called before starting streams or after stopping +// streams. void snd_motu_stream_destroy_duplex(struct snd_motu *motu) { - destroy_stream(motu, AMDTP_IN_STREAM); - destroy_stream(motu, AMDTP_OUT_STREAM); + amdtp_domain_destroy(&motu->domain); + + destroy_stream(motu, &motu->rx_stream); + destroy_stream(motu, &motu->tx_stream); motu->substreams_counter = 0; } diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 03cda2166ea3..72908b4de77c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -247,6 +247,17 @@ static const struct snd_motu_spec motu_audio_express = { .analog_out_ports = 4, }; +static const struct snd_motu_spec motu_4pre = { + .name = "4pre", + .protocol = &snd_motu_protocol_v3, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_TX_MICINST_CHUNK | + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN, + .analog_in_ports = 2, + .analog_out_ports = 2, +}; + #define SND_MOTU_DEV_ENTRY(model, data) \ { \ .match_flags = IEEE1394_MATCH_VENDOR_ID | \ @@ -265,6 +276,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), + SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre), { } }; MODULE_DEVICE_TABLE(ieee1394, motu_id_table); diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 09d1451d7de4..350ee2c16f4a 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -69,6 +69,8 @@ struct snd_motu { int dev_lock_count; bool dev_lock_changed; wait_queue_head_t hwdep_wait; + + struct amdtp_domain domain; }; enum snd_motu_spec_flags { diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9ea39348cdf5..7c6d1c277d4d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(hw_params); mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, rate, channels); if (err >= 0) ++oxfw->substreams_count; diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 74c972d25c66..3c9a796b6526 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -114,19 +114,13 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) if (err < 0) return err; - err = amdtp_stream_start(stream, conn->resources.channel, conn->speed); + err = amdtp_domain_add_stream(&oxfw->domain, stream, + conn->resources.channel, conn->speed); if (err < 0) { cmp_connection_break(conn); return err; } - // Wait first packet. - if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(stream); - cmp_connection_break(conn); - return -ETIMEDOUT; - } - return 0; } @@ -280,12 +274,12 @@ int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, pcm_channels = formation.pcm; } if (formation.rate != rate || formation.pcm != pcm_channels) { - amdtp_stream_stop(&oxfw->rx_stream); + amdtp_domain_stop(&oxfw->domain); + cmp_connection_break(&oxfw->in_conn); cmp_connection_release(&oxfw->in_conn); if (oxfw->has_output) { - amdtp_stream_stop(&oxfw->tx_stream); cmp_connection_break(&oxfw->out_conn); cmp_connection_release(&oxfw->out_conn); } @@ -325,30 +319,46 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) if (amdtp_streaming_error(&oxfw->rx_stream) || amdtp_streaming_error(&oxfw->tx_stream)) { - amdtp_stream_stop(&oxfw->rx_stream); - cmp_connection_break(&oxfw->in_conn); + amdtp_domain_stop(&oxfw->domain); - if (oxfw->has_output) { - amdtp_stream_stop(&oxfw->tx_stream); + cmp_connection_break(&oxfw->in_conn); + if (oxfw->has_output) cmp_connection_break(&oxfw->out_conn); - } } if (!amdtp_stream_running(&oxfw->rx_stream)) { err = start_stream(oxfw, &oxfw->rx_stream); if (err < 0) { dev_err(&oxfw->unit->device, - "fail to start rx stream: %d\n", err); + "fail to prepare rx stream: %d\n", err); goto error; } - } - if (oxfw->has_output) { - if (!amdtp_stream_running(&oxfw->tx_stream)) { + if (oxfw->has_output && + !amdtp_stream_running(&oxfw->tx_stream)) { err = start_stream(oxfw, &oxfw->tx_stream); if (err < 0) { dev_err(&oxfw->unit->device, - "fail to start tx stream: %d\n", err); + "fail to prepare tx stream: %d\n", err); + goto error; + } + } + + err = amdtp_domain_start(&oxfw->domain); + if (err < 0) + goto error; + + // Wait first packet. + if (!amdtp_stream_wait_callback(&oxfw->rx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + + if (oxfw->has_output) { + if (!amdtp_stream_wait_callback(&oxfw->tx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; goto error; } } @@ -356,24 +366,24 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) return 0; error: - amdtp_stream_stop(&oxfw->rx_stream); + amdtp_domain_stop(&oxfw->domain); + cmp_connection_break(&oxfw->in_conn); - if (oxfw->has_output) { - amdtp_stream_stop(&oxfw->tx_stream); + if (oxfw->has_output) cmp_connection_break(&oxfw->out_conn); - } + return err; } void snd_oxfw_stream_stop_duplex(struct snd_oxfw *oxfw) { if (oxfw->substreams_count == 0) { - amdtp_stream_stop(&oxfw->rx_stream); + amdtp_domain_stop(&oxfw->domain); + cmp_connection_break(&oxfw->in_conn); cmp_connection_release(&oxfw->in_conn); if (oxfw->has_output) { - amdtp_stream_stop(&oxfw->tx_stream); cmp_connection_break(&oxfw->out_conn); cmp_connection_release(&oxfw->out_conn); } @@ -409,13 +419,22 @@ int snd_oxfw_stream_init_duplex(struct snd_oxfw *oxfw) } } - return 0; + err = amdtp_domain_init(&oxfw->domain); + if (err < 0) { + destroy_stream(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + destroy_stream(oxfw, &oxfw->tx_stream); + } + + return err; } // This function should be called before starting the stream or after stopping // the streams. void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw) { + amdtp_domain_destroy(&oxfw->domain); + destroy_stream(oxfw, &oxfw->rx_stream); if (oxfw->has_output) @@ -424,13 +443,13 @@ void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw) void snd_oxfw_stream_update_duplex(struct snd_oxfw *oxfw) { - amdtp_stream_stop(&oxfw->rx_stream); + amdtp_domain_stop(&oxfw->domain); + cmp_connection_break(&oxfw->in_conn); amdtp_stream_pcm_abort(&oxfw->rx_stream); if (oxfw->has_output) { - amdtp_stream_stop(&oxfw->tx_stream); cmp_connection_break(&oxfw->out_conn); amdtp_stream_pcm_abort(&oxfw->tx_stream); diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index cb69ab87bb14..c9627b8c5d6e 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -63,6 +63,8 @@ struct snd_oxfw { const struct ieee1394_device_id *entry; void *spec; + + struct amdtp_domain domain; }; /* diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index 0d35359d25cd..0ecafd0c6722 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit, packets_per_page = PAGE_SIZE / packet_size; if (WARN_ON(!packets_per_page)) { err = -EINVAL; - goto error; + goto err_packets; } pages = DIV_ROUND_UP(count, packets_per_page); diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index 95fb10b7a737..e80bb84c43f6 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -32,19 +32,24 @@ int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate) return amdtp_stream_set_parameters(s, rate, data_channels); } -static void write_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) +static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_tscm *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; const u32 *src; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; for (i = 0; i < frames; ++i) { for (c = 0; c < channels; ++c) { @@ -57,19 +62,24 @@ static void write_pcm_s32(struct amdtp_stream *s, } } -static void read_pcm_s32(struct amdtp_stream *s, - struct snd_pcm_substream *pcm, - __be32 *buffer, unsigned int frames) +static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm, + __be32 *buffer, unsigned int frames, + unsigned int pcm_frames) { struct amdtp_tscm *p = s->protocol; + unsigned int channels = p->pcm_channels; struct snd_pcm_runtime *runtime = pcm->runtime; - unsigned int channels, remaining_frames, i, c; + unsigned int pcm_buffer_pointer; + int remaining_frames; u32 *dst; + int i, c; + + pcm_buffer_pointer = s->pcm_buffer_pointer + pcm_frames; + pcm_buffer_pointer %= runtime->buffer_size; - 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; + frames_to_bytes(runtime, pcm_buffer_pointer); + remaining_frames = runtime->buffer_size - pcm_buffer_pointer; /* The first data channel is for event counter. */ buffer += 1; @@ -165,65 +175,82 @@ static void read_status_messages(struct amdtp_stream *s, } } -static unsigned int process_tx_data_blocks(struct amdtp_stream *s, - __be32 *buffer, - unsigned int data_blocks, - unsigned int *syt) +static unsigned int process_ir_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm; + unsigned int pcm_frames = 0; + int i; + + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - pcm = READ_ONCE(s->pcm); - if (data_blocks > 0 && pcm) - read_pcm_s32(s, pcm, buffer, data_blocks); + if (pcm) { + read_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } - read_status_messages(s, buffer, data_blocks); + read_status_messages(s, buf, data_blocks); + } - return 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) +static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, + const struct pkt_desc *descs, + unsigned int packets, + struct snd_pcm_substream *pcm) { - struct snd_pcm_substream *pcm; + unsigned int pcm_frames = 0; + int i; - /* This field is not used. */ - *syt = 0x0000; + for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = descs + i; + __be32 *buf = desc->ctx_payload; + unsigned int data_blocks = desc->data_blocks; - pcm = READ_ONCE(s->pcm); - if (pcm) - write_pcm_s32(s, pcm, buffer, data_blocks); - else - write_pcm_silence(s, buffer, data_blocks); + if (pcm) { + write_pcm_s32(s, pcm, buf, data_blocks, pcm_frames); + pcm_frames += data_blocks; + } else { + write_pcm_silence(s, buf, data_blocks); + } + } - return data_blocks; + return pcm_frames; } int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, unsigned int pcm_channels) { - amdtp_stream_process_data_blocks_t process_data_blocks; + amdtp_stream_process_ctx_payloads_t process_ctx_payloads; struct amdtp_tscm *p; unsigned int fmt; int err; if (dir == AMDTP_IN_STREAM) { fmt = AMDTP_FMT_TSCM_TX; - process_data_blocks = process_tx_data_blocks; + process_ctx_payloads = process_ir_ctx_payloads; } else { fmt = AMDTP_FMT_TSCM_RX; - process_data_blocks = process_rx_data_blocks; + process_ctx_payloads = process_it_ctx_payloads; } err = amdtp_stream_init(s, unit, dir, - CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt, - process_data_blocks, sizeof(struct amdtp_tscm)); + CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt, + process_ctx_payloads, sizeof(struct amdtp_tscm)); if (err < 0) return 0; - /* Use fixed value for FDF field. */ - s->ctx_data.rx.fdf = 0x00; + if (dir == AMDTP_OUT_STREAM) { + // Use fixed value for FDF field. + s->ctx_data.rx.fdf = 0x00; + // Not used. + s->ctx_data.rx.syt_override = 0x0000; + } /* This protocol uses fixed number of data channels for PCM samples. */ p = s->protocol; diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index b5ced5415e40..2377732caa52 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -56,6 +56,9 @@ static int pcm_open(struct snd_pcm_substream *substream) goto err_locked; err = snd_tscm_stream_get_clock(tscm, &clock); + if (err < 0) + goto err_locked; + if (clock != SND_TSCM_CLOCK_INTERNAL || amdtp_stream_pcm_running(&tscm->rx_stream) || amdtp_stream_pcm_running(&tscm->tx_stream)) { diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index e852e46ebe6f..adf69a520b80 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -8,20 +8,37 @@ #include <linux/delay.h> #include "tascam.h" +#define CLOCK_STATUS_MASK 0xffff0000 +#define CLOCK_CONFIG_MASK 0x0000ffff + #define CALLBACK_TIMEOUT 500 static int get_clock(struct snd_tscm *tscm, u32 *data) { + int trial = 0; __be32 reg; int err; - err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, - TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, - ®, sizeof(reg), 0); - if (err >= 0) + while (trial++ < 5) { + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, + ®, sizeof(reg), 0); + if (err < 0) + return err; + *data = be32_to_cpu(reg); + if (*data & CLOCK_STATUS_MASK) + break; - return err; + // In intermediate state after changing clock status. + msleep(50); + } + + // Still in the intermediate state. + if (trial >= 5) + return -EAGAIN; + + return 0; } static int set_clock(struct snd_tscm *tscm, unsigned int rate, @@ -34,7 +51,7 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate, err = get_clock(tscm, &data); if (err < 0) return err; - data &= 0x0000ffff; + data &= CLOCK_CONFIG_MASK; if (rate > 0) { data &= 0x000000ff; @@ -79,17 +96,14 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate, int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate) { - u32 data = 0x0; - unsigned int trials = 0; + u32 data; int err; - while (data == 0x0 || trials++ < 5) { - err = get_clock(tscm, &data); - if (err < 0) - return err; + err = get_clock(tscm, &data); + if (err < 0) + return err; - data = (data & 0xff000000) >> 24; - } + data = (data & 0xff000000) >> 24; /* Check base rate. */ if ((data & 0x0f) == 0x01) @@ -180,9 +194,6 @@ static void finish_session(struct snd_tscm *tscm) { __be32 reg; - amdtp_stream_stop(&tscm->rx_stream); - amdtp_stream_stop(&tscm->tx_stream); - reg = 0; snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING, @@ -287,38 +298,68 @@ static int keep_resources(struct snd_tscm *tscm, unsigned int rate, fw_parent_device(tscm->unit)->max_speed); } -int snd_tscm_stream_init_duplex(struct snd_tscm *tscm) +static int init_stream(struct snd_tscm *tscm, struct amdtp_stream *s) { + struct fw_iso_resources *resources; + enum amdtp_stream_direction dir; unsigned int pcm_channels; int err; - /* For out-stream. */ - err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit); - if (err < 0) - return err; - pcm_channels = tscm->spec->pcm_playback_analog_channels; + if (s == &tscm->tx_stream) { + resources = &tscm->tx_resources; + dir = AMDTP_IN_STREAM; + pcm_channels = tscm->spec->pcm_capture_analog_channels; + } else { + resources = &tscm->rx_resources; + dir = AMDTP_OUT_STREAM; + pcm_channels = tscm->spec->pcm_playback_analog_channels; + } + if (tscm->spec->has_adat) pcm_channels += 8; if (tscm->spec->has_spdif) pcm_channels += 2; - err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM, - pcm_channels); + + err = fw_iso_resources_init(resources, tscm->unit); if (err < 0) return err; - /* For in-stream. */ - err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit); + err = amdtp_tscm_init(s, tscm->unit, dir, pcm_channels); if (err < 0) - return err; - pcm_channels = tscm->spec->pcm_capture_analog_channels; - if (tscm->spec->has_adat) - pcm_channels += 8; - if (tscm->spec->has_spdif) - pcm_channels += 2; - err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM, - pcm_channels); + fw_iso_resources_free(resources); + + return err; +} + +static void destroy_stream(struct snd_tscm *tscm, struct amdtp_stream *s) +{ + amdtp_stream_destroy(s); + + if (s == &tscm->tx_stream) + fw_iso_resources_destroy(&tscm->tx_resources); + else + fw_iso_resources_destroy(&tscm->rx_resources); +} + +int snd_tscm_stream_init_duplex(struct snd_tscm *tscm) +{ + int err; + + err = init_stream(tscm, &tscm->tx_stream); if (err < 0) - amdtp_stream_destroy(&tscm->rx_stream); + return err; + + err = init_stream(tscm, &tscm->rx_stream); + if (err < 0) { + destroy_stream(tscm, &tscm->tx_stream); + return err; + } + + err = amdtp_domain_init(&tscm->domain); + if (err < 0) { + destroy_stream(tscm, &tscm->tx_stream); + destroy_stream(tscm, &tscm->rx_stream); + } return err; } @@ -326,24 +367,20 @@ int snd_tscm_stream_init_duplex(struct snd_tscm *tscm) // At bus reset, streaming is stopped and some registers are clear. void snd_tscm_stream_update_duplex(struct snd_tscm *tscm) { - amdtp_stream_pcm_abort(&tscm->tx_stream); - amdtp_stream_stop(&tscm->tx_stream); + amdtp_domain_stop(&tscm->domain); + amdtp_stream_pcm_abort(&tscm->tx_stream); amdtp_stream_pcm_abort(&tscm->rx_stream); - amdtp_stream_stop(&tscm->rx_stream); } -/* - * This function should be called before starting streams or after stopping - * streams. - */ +// This function should be called before starting streams or after stopping +// streams. void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm) { - amdtp_stream_destroy(&tscm->rx_stream); - amdtp_stream_destroy(&tscm->tx_stream); + amdtp_domain_destroy(&tscm->domain); - fw_iso_resources_destroy(&tscm->rx_resources); - fw_iso_resources_destroy(&tscm->tx_resources); + destroy_stream(tscm, &tscm->rx_stream); + destroy_stream(tscm, &tscm->tx_stream); } int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate) @@ -356,6 +393,8 @@ int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate) return err; if (tscm->substreams_counter == 0 || rate != curr_rate) { + amdtp_domain_stop(&tscm->domain); + finish_session(tscm); fw_iso_resources_free(&tscm->tx_resources); @@ -388,8 +427,10 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; if (amdtp_streaming_error(&tscm->rx_stream) || - amdtp_streaming_error(&tscm->tx_stream)) + amdtp_streaming_error(&tscm->tx_stream)) { + amdtp_domain_stop(&tscm->domain); finish_session(tscm); + } if (generation != fw_parent_device(tscm->unit)->card->generation) { err = fw_iso_resources_update(&tscm->tx_resources); @@ -402,6 +443,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) } if (!amdtp_stream_running(&tscm->rx_stream)) { + int spd = fw_parent_device(tscm->unit)->max_speed; + err = set_stream_formats(tscm, rate); if (err < 0) goto error; @@ -410,27 +453,23 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_stream_start(&tscm->rx_stream, - tscm->rx_resources.channel, - fw_parent_device(tscm->unit)->max_speed); + err = amdtp_domain_add_stream(&tscm->domain, &tscm->rx_stream, + tscm->rx_resources.channel, spd); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->rx_stream, - CALLBACK_TIMEOUT)) { - err = -ETIMEDOUT; + err = amdtp_domain_add_stream(&tscm->domain, &tscm->tx_stream, + tscm->tx_resources.channel, spd); + if (err < 0) goto error; - } - } - if (!amdtp_stream_running(&tscm->tx_stream)) { - err = amdtp_stream_start(&tscm->tx_stream, - tscm->tx_resources.channel, - fw_parent_device(tscm->unit)->max_speed); + err = amdtp_domain_start(&tscm->domain); if (err < 0) - goto error; + return err; - if (!amdtp_stream_wait_callback(&tscm->tx_stream, + if (!amdtp_stream_wait_callback(&tscm->rx_stream, + CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(&tscm->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -439,6 +478,7 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; error: + amdtp_domain_stop(&tscm->domain); finish_session(tscm); return err; @@ -447,6 +487,7 @@ error: void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm) { if (tscm->substreams_counter == 0) { + amdtp_domain_stop(&tscm->domain); finish_session(tscm); fw_iso_resources_free(&tscm->tx_resources); diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 231052db5680..addc464503bc 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,6 +39,9 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. }; static int identify_model(struct snd_tscm *tscm) @@ -214,7 +217,6 @@ static const struct ieee1394_device_id snd_tscm_id_table[] = { .vendor_id = 0x00022e, .specifier_id = 0x00022e, }, - /* FE-08 requires reverse-engineering because it just has faders. */ {} }; MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 734e5bb9c3da..15bd335fa07f 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -97,6 +97,8 @@ struct snd_tscm { struct snd_firewire_tascam_change queue[SND_TSCM_QUEUE_COUNT]; unsigned int pull_pos; unsigned int push_pos; + + struct amdtp_domain domain; }; #define TSCM_ADDR_BASE 0xffff00000000ull @@ -127,6 +129,26 @@ struct snd_tscm { #define TSCM_OFFSET_MIDI_RX_QUAD 0x4000 +// Although FE-8 supports the above registers, it has no I/O interfaces for +// audio samples and music messages. Otherwise it supports another notification +// for status and control message as well as LED brightening. The message +// consists of quadlet-aligned data up to 32 quadlets. The first byte of message +// is fixed to 0x40. The second byte is between 0x00 to 0x1f and represent each +// control: +// fader: 0x00-0x07 +// button: 0x0d, 0x0e +// knob: 0x14-0x1b +// sensing: 0x0b +// +// The rest two bytes represent state of the controls; e.g. current value for +// fader and knob, bitmasks for button and sensing. +// Just after turning on, 32 quadlets messages with 0x00-0x1f are immediately +// sent in one transaction. After, several quadlets are sent in one transaction. +// +// TSCM_OFFSET_FE8_CTL_TX_ON 0x0310 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_HI 0x0314 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_LO 0x0318 + enum snd_tscm_clock { SND_TSCM_CLOCK_INTERNAL = 0, SND_TSCM_CLOCK_WORD = 1, diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -271,6 +271,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); + /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery */ diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index cd25e2b3f7f2..8f19876244eb 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -9,8 +9,11 @@ #include <linux/module.h> #include <linux/export.h> #include <sound/hdaudio.h> +#include "local.h" #include "trace.h" +static void snd_hdac_bus_process_unsol_events(struct work_struct *work); + static const struct hdac_bus_ops default_ops = { .command = snd_hdac_bus_send_cmd, .get_response = snd_hdac_bus_get_response, @@ -148,7 +151,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_queue_event); /* * process queued unsolicited events */ -void snd_hdac_bus_process_unsol_events(struct work_struct *work) +static void snd_hdac_bus_process_unsol_events(struct work_struct *work) { struct hdac_bus *bus = container_of(work, struct hdac_bus, unsol_work); struct hdac_device *codec; @@ -171,7 +174,6 @@ void snd_hdac_bus_process_unsol_events(struct work_struct *work) drv->unsol_event(codec, res); } } -EXPORT_SYMBOL_GPL(snd_hdac_bus_process_unsol_events); /** * snd_hdac_bus_add_device - Add a codec to bus @@ -196,7 +198,6 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) bus->num_codecs++; return 0; } -EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device); /** * snd_hdac_bus_remove_device - Remove a codec from bus @@ -215,7 +216,6 @@ void snd_hdac_bus_remove_device(struct hdac_bus *bus, bus->num_codecs--; flush_work(&bus->unsol_work); } -EXPORT_SYMBOL_GPL(snd_hdac_bus_remove_device); #ifdef CONFIG_SND_HDA_ALIGNED_MMIO /* Helpers for aligned read/write of mmio space, for Tegra */ diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7e7be8e4dcf9..d3999e7b0705 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,6 +447,8 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); + synchronize_irq(bus->irq); + /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..9f3e37511408 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -218,8 +218,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); * * Return an encoded command verb or -1 for error. */ -unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, - unsigned int verb, unsigned int parm) +static unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int parm) { u32 val, addr; @@ -237,7 +237,6 @@ unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, val |= parm; return val; } -EXPORT_SYMBOL_GPL(snd_hdac_make_cmd); /** * snd_hdac_exec_verb - execute an encoded verb @@ -258,7 +257,6 @@ int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd, return codec->exec_verb(codec, cmd, flags, res); return snd_hdac_bus_exec_verb(codec->bus, codec->addr, cmd, res); } -EXPORT_SYMBOL_GPL(snd_hdac_exec_verb); /** diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 1192c7561d62..3c2db3816029 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -136,10 +136,12 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!acomp) return -ENODEV; if (!acomp->ops) { - request_module("i915"); - /* 60s timeout */ - wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(60 * 1000)); + if (!IS_ENABLED(CONFIG_MODULES) || + !request_module("i915")) { + /* 60s timeout */ + wait_for_completion_timeout(&bind_complete, + msecs_to_jiffies(60 * 1000)); + } } if (!acomp->ops) { dev_info(bus->dev, "couldn't bind with audio component\n"); diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index f399a1552e73..286361ecd640 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -21,6 +21,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_regmap.h> +#include "local.h" static int codec_pm_lock(struct hdac_device *codec) { diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index fc68d4ce0a37..d8fe7ff0cd58 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -229,11 +229,7 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev) /* set the interrupt enable bits in the descriptor control register */ snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_INT_MASK); - if (azx_dev->direction == SNDRV_PCM_STREAM_PLAYBACK) - azx_dev->fifo_size = - snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1; - else - azx_dev->fifo_size = 0; + azx_dev->fifo_size = snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1; /* when LPIB delay correction gives a small negative value, * we ignore it; currently set the threshold statically to diff --git a/sound/hda/local.h b/sound/hda/local.h index 877631e39373..5b935219352f 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -33,4 +33,11 @@ int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid, int num_nodes); void hda_widget_sysfs_exit(struct hdac_device *codec); +int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec); +void snd_hdac_bus_remove_device(struct hdac_bus *bus, + struct hdac_device *codec); + +int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd, + unsigned int flags, unsigned int *res); + #endif /* __HDAC_LOCAL_H */ diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 5f59316f982a..7d15093844b9 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -775,11 +775,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak) return err; memset(&knew, 0, sizeof(knew)); - knew.name = ak->adc_info[mixer_ch].selector_name; - if (!knew.name) { + if (!ak->adc_info || + !ak->adc_info[mixer_ch].selector_name) { knew.name = "Capture Channel"; knew.index = mixer_ch + ak->idx_offset * 2; - } + } else + knew.name = ak->adc_info[mixer_ch].selector_name; knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; knew.info = ak4xxx_capture_source_info; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 162338f1b68a..ff031d670400 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -80,7 +80,7 @@ int snd_sbdsp_reset(struct snd_sb *chip) static int snd_sbdsp_version(struct snd_sb * chip) { - unsigned int result = -ENODEV; + unsigned int result; snd_sbdsp_command(chip, SB_DSP_GET_VERSION); result = (short) snd_sbdsp_get_byte(chip) << 8; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index aec1c46e6697..c5b1d5900eed 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -788,7 +788,6 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) dev->patch_status[header->number] |= WF_SLOT_FILLED; - bptr = buf; bptr = munge_int32 (header->number, buf, 2); munge_buf ((unsigned char *)&header->hdr.p, bptr, WF_PATCH_BYTES); diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 83653683fd68..823ccfa089b2 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1432,25 +1432,25 @@ static int FalconMixerIoctl(u_int cmd, u_long arg) { int data; switch (cmd) { - case SOUND_MIXER_READ_RECMASK: + case SOUND_MIXER_READ_RECMASK: return IOCTL_OUT(arg, SOUND_MASK_MIC); - case SOUND_MIXER_READ_DEVMASK: + case SOUND_MIXER_READ_DEVMASK: return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC | SOUND_MASK_SPEAKER); - case SOUND_MIXER_READ_STEREODEVS: + case SOUND_MIXER_READ_STEREODEVS: return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC); - case SOUND_MIXER_READ_VOLUME: + case SOUND_MIXER_READ_VOLUME: return IOCTL_OUT(arg, VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) | VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8); - case SOUND_MIXER_READ_CAPS: + case SOUND_MIXER_READ_CAPS: return IOCTL_OUT(arg, SOUND_CAP_EXCL_INPUT); - case SOUND_MIXER_WRITE_MIC: + case SOUND_MIXER_WRITE_MIC: IOCTL_IN(arg, data); tt_dmasnd.input_gain = RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); - /* fall thru, return set value */ - case SOUND_MIXER_READ_MIC: + /* fall through - return set value */ + case SOUND_MIXER_READ_MIC: return IOCTL_OUT(arg, RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain & 0xf) << 8); diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 96b4601aae73..66f6c3bf08e3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -596,11 +596,6 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, return err; } -static const struct snd_kcontrol_new snd_ac97_controls_master_mono[2] = { -AC97_SINGLE("Master Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -AC97_SINGLE("Master Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1) -}; - static const struct snd_kcontrol_new snd_ac97_controls_tone[2] = { AC97_SINGLE("Tone Control - Bass", AC97_MASTER_TONE, 8, 15, 1), AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..ca9125726be2 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage = chip->comm_page; + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; - commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..2c6d2becfe1a 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -824,6 +824,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; + if (++depth > 10) + break; if (fix->chained_before) apply_fixup(codec, fix->chain_id, action, depth + 1); @@ -863,8 +865,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } if (!fix->chained || fix->chained_before) break; - if (++depth > 10) - break; id = fix->chain_id; } } @@ -884,7 +884,8 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); #define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) static bool pin_config_match(struct hda_codec *codec, - const struct hda_pintbl *pins) + const struct hda_pintbl *pins, + bool match_all_pins) { const struct hda_pincfg *pin; int i; @@ -908,7 +909,8 @@ static bool pin_config_match(struct hda_codec *codec, return false; } } - if (!found && (cfg & 0xf0000000) != 0x40000000) + if (match_all_pins && + !found && (cfg & 0xf0000000) != 0x40000000) return false; } @@ -920,10 +922,12 @@ static bool pin_config_match(struct hda_codec *codec, * @codec: the HDA codec * @pin_quirk: zero-terminated pin quirk list * @fixlist: the fixup list + * @match_all_pins: all valid pins must match with the table entries */ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist) + const struct hda_fixup *fixlist, + bool match_all_pins) { const struct snd_hda_pin_quirk *pq; @@ -935,7 +939,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins, match_all_pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 133200d31170..a2fb19129219 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2948,7 +2948,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume; + bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ee5504e2441f..6387c7e90918 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; @@ -615,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); + /* by some reason, the playback stream stalls on PulseAudio with + * tsched=1 when a capture stream triggers. Until we figure out the + * real cause, disable tsched mode by telling the PCM info flag. + */ + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) + runtime->hw.info |= SNDRV_PCM_INFO_BATCH; + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory @@ -789,6 +794,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, unsigned long timeout; unsigned long loopcounter; int do_poll = 0; + bool warned = false; again: timeout = jiffies + msecs_to_jiffies(1000); @@ -808,9 +814,17 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, spin_unlock_irq(&bus->reg_lock); if (time_after(jiffies, timeout)) break; - if (hbus->needs_damn_long_delay || loopcounter > 3000) +#define LOOP_COUNT_MAX 3000 + if (hbus->needs_damn_long_delay || + loopcounter > LOOP_COUNT_MAX) { + if (loopcounter > LOOP_COUNT_MAX && !warned) { + dev_dbg_ratelimited(chip->card->dev, + "too slow response, last cmd=%#08x\n", + bus->last_cmd[addr]); + warned = true; + } msleep(2); /* temporary workaround */ - else { + } else { udelay(10); cond_resched(); } @@ -864,10 +878,13 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, */ if (hbus->allow_bus_reset && !hbus->response_reset && !hbus->in_reset) { hbus->response_reset = 1; + dev_err(chip->card->dev, + "No response from codec, resetting bus: last cmd=0x%08x\n", + bus->last_cmd[addr]); return -EAGAIN; /* give a chance to retry */ } - dev_err(chip->card->dev, + dev_WARN(chip->card->dev, "azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n", bus->last_cmd[addr]); chip->single_cmd = 1; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 146a71e0d594..82e26442724b 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -31,7 +31,7 @@ /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -/* 17 unused */ +#define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 485edaba0037..10d502328b76 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6009,7 +6009,8 @@ int snd_hda_gen_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); - snd_hda_apply_verbs(codec); + if (!spec->skip_verbs) + snd_hda_apply_verbs(codec); init_multi_out(codec); init_extra_out(codec); @@ -6051,6 +6052,24 @@ void snd_hda_gen_free(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_gen_free); +/** + * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting + * @codec: the HDA codec + * + * This can be put as patch_ops reboot_notify function. + */ +void snd_hda_gen_reboot_notify(struct hda_codec *codec) +{ + /* Make the codec enter D3 to avoid spurious noises from the internal + * speaker during (and after) reboot + */ + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify); + #ifdef CONFIG_PM /** * snd_hda_gen_check_power_status - check the loopback power save state @@ -6078,6 +6097,7 @@ static const struct hda_codec_ops generic_patch_ops = { .init = snd_hda_gen_init, .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, + .reboot_notify = snd_hda_gen_reboot_notify, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, #endif @@ -6100,7 +6120,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); if (err < 0) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 35a670a71c42..fb9f1a90238b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -243,6 +243,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */ /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; @@ -332,6 +333,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, struct auto_pin_cfg *cfg); int snd_hda_gen_build_controls(struct hda_codec *codec); int snd_hda_gen_build_pcms(struct hda_codec *codec); +void snd_hda_gen_reboot_notify(struct hda_codec *codec); /* standard jack event callbacks */ void snd_hda_gen_hp_automute(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6963dd852b5b..240f4ca76391 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -65,6 +65,7 @@ enum { POS_FIX_VIACOMBO, POS_FIX_COMBO, POS_FIX_SKL, + POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -84,8 +85,6 @@ enum { #define INTEL_SCH_HDA_DEVC 0x78 #define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) -/* Define IN stream 0 FIFO size offset in VIA controller */ -#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90 /* Define VIA HD Audio Device ID*/ #define VIA_HDAC_DEVICE_ID 0x3288 @@ -137,7 +136,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -270,6 +269,7 @@ enum { AZX_DRIVER_CTX, AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, + AZX_DRIVER_ZHAOXIN, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -317,11 +317,10 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) -#define AZX_DCAPS_INTEL_BROXTON \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) +#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -337,6 +336,11 @@ enum { #define AZX_DCAPS_PRESET_ATI_HDMI_NS \ (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) +/* quirks for AMD SB */ +#define AZX_DCAPS_PRESET_AMD_SB \ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ @@ -352,7 +356,7 @@ enum { */ #ifdef SUPPORT_VGA_SWITCHEROO #define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) -#define needs_eld_notify_link(chip) ((chip)->need_eld_notify_link) +#define needs_eld_notify_link(chip) ((chip)->bus.keep_power) #else #define use_vga_switcheroo(chip) 0 #define needs_eld_notify_link(chip) false @@ -384,6 +388,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", + [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -810,11 +815,7 @@ static unsigned int azx_via_get_position(struct azx *chip, mod_dma_pos = le32_to_cpu(*azx_dev->core.posbuf); mod_dma_pos %= azx_dev->core.period_bytes; - /* azx_dev->fifo_size can't get FIFO size of in stream. - * Get from base address + offset. - */ - fifo_size = readw(azx_bus(chip)->remap_addr + - VIA_IN_STREAM0_FIFO_SIZE_OFFSET); + fifo_size = azx_stream(azx_dev)->fifo_size - 1; if (azx_dev->insufficient) { /* Link position never gather than FIFO size */ @@ -846,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip, return bound_pos + mod_dma_pos; } +#define AMD_FIFO_SIZE 32 + +/* get the current DMA position with FIFO size correction */ +static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int pos, delay; + + pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); + if (!runtime) + return pos; + + runtime->delay = AMD_FIFO_SIZE; + delay = frames_to_bytes(runtime, AMD_FIFO_SIZE); + if (azx_dev->insufficient) { + if (pos < delay) { + delay = pos; + runtime->delay = bytes_to_frames(runtime, pos); + } else { + azx_dev->insufficient = 0; + } + } + + /* correct the DMA position for capture stream */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (pos < delay) + pos += azx_dev->core.bufsize; + pos -= delay; + } + + return pos; +} + +static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + + /* just read back the calculated value in the above */ + return substream->runtime->delay; +} + static unsigned int azx_skl_get_dpib_pos(struct azx *chip, struct azx_dev *azx_dev) { @@ -1101,7 +1145,7 @@ static int azx_runtime_idle(struct device *dev) return -EBUSY; /* ELD notification gets broken when HD-audio bus is off */ - if (needs_eld_notify_link(hda)) + if (needs_eld_notify_link(chip)) return -EBUSY; return 0; @@ -1212,7 +1256,7 @@ static void setup_vga_switcheroo_runtime_pm(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hda_codec *codec; - if (hda->use_vga_switcheroo && !hda->need_eld_notify_link) { + if (hda->use_vga_switcheroo && !needs_eld_notify_link(chip)) { list_for_each_codec(codec, &chip->bus) codec->auto_runtime_pm = 1; /* reset the power save setup */ @@ -1226,10 +1270,9 @@ static void azx_vs_gpu_bound(struct pci_dev *pci, { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); if (client_id == VGA_SWITCHEROO_DIS) - hda->need_eld_notify_link = 0; + chip->bus.keep_power = 0; setup_vga_switcheroo_runtime_pm(chip); } @@ -1241,7 +1284,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; - hda->need_eld_notify_link = 1; /* cleared in gpu_bound op */ + chip->bus.keep_power = 1; /* cleared in either gpu_bound op or codec probe */ chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } @@ -1305,9 +1348,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { + azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); - azx_stop_chip(chip); } if (bus->irq >= 0) @@ -1422,6 +1465,7 @@ static int check_position_fix(struct azx *chip, int fix) case POS_FIX_VIACOMBO: case POS_FIX_COMBO: case POS_FIX_SKL: + case POS_FIX_FIFO: return fix; } @@ -1438,6 +1482,10 @@ static int check_position_fix(struct azx *chip, int fix) dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n"); return POS_FIX_VIACOMBO; } + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) { + dev_dbg(chip->card->dev, "Using FIFO position fix\n"); + return POS_FIX_FIFO; + } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { dev_dbg(chip->card->dev, "Using LPIB position fix\n"); return POS_FIX_LPIB; @@ -1458,6 +1506,7 @@ static void assign_position_fix(struct azx *chip, int fix) [POS_FIX_VIACOMBO] = azx_via_get_position, [POS_FIX_COMBO] = azx_get_pos_lpib, [POS_FIX_SKL] = azx_get_pos_skl, + [POS_FIX_FIFO] = azx_get_pos_fifo, }; chip->get_position[0] = chip->get_position[1] = callbacks[fix]; @@ -1472,6 +1521,9 @@ static void assign_position_fix(struct azx *chip, int fix) azx_get_delay_from_lpib; } + if (fix == POS_FIX_FIFO) + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_fifo; } /* @@ -2421,14 +2473,19 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD, X370 & co */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, + /* AMD, X570 & co */ + { PCI_DEVICE(0x1022, 0x1487), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, @@ -2563,6 +2620,8 @@ static const struct pci_device_id azx_ids[] = { .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, + /* Zhaoxin */ + { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 1468865e0342..2acfff3da1a0 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -25,7 +25,6 @@ struct hda_intel { /* vga_switcheroo setup */ unsigned int use_vga_switcheroo:1; - unsigned int need_eld_notify_link:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 349a8312d06a..3942e1b528d8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -361,7 +361,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_fixup *fixlist); void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist); + const struct hda_fixup *fixlist, + bool match_all_pins); /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e283966bdbb1..bc9dd8e6fd86 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -357,6 +357,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC), SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0d51823d7270..6d1fb7c11f17 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f8d0845ee1e..968d3caab6ac 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -163,23 +163,10 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - switch (codec->core.vendor_id) { - case 0x14f12008: /* CX8200 */ - case 0x14f150f2: /* CX20722 */ - case 0x14f150f4: /* CX20724 */ - break; - default: - return; - } - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); - - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); + snd_hda_gen_reboot_notify(codec); } static void cx_auto_free(struct hda_codec *codec) @@ -624,18 +611,20 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, /* update LED status via GPIO */ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + bool led_on) { struct conexant_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; if (spec->mute_led_polarity) - enabled = !enabled; + led_on = !led_on; - if (enabled) - spec->gpio_led &= ~mask; - else + if (led_on) spec->gpio_led |= mask; + else + spec->gpio_led &= ~mask; + codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n", + mask, led_on, spec->gpio_led); if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); @@ -646,8 +635,8 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct conexant_spec *spec = codec->spec; - - cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + /* muted -> LED on */ + cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -669,7 +658,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, {} }; - codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; @@ -1083,6 +1071,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bea7b0961080..795cbda32cbb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -18,6 +18,7 @@ #include <linux/init.h> #include <linux/delay.h> +#include <linux/pci.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/pm_runtime.h> @@ -119,6 +120,7 @@ struct hdmi_pcm { }; struct hdmi_spec { + struct hda_codec *codec; int num_cvts; struct snd_array cvts; /* struct hdmi_spec_per_cvt */ hda_nid_t cvt_nids[4]; /* only for haswell fix */ @@ -163,9 +165,11 @@ struct hdmi_spec { struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; - /* i915/powerwell (Haswell+/Valleyview+) specific */ - bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ + bool use_jack_detect; /* jack detection enabled */ + bool use_acomp_notifier; /* use eld_notify callback for hotplug */ + bool acomp_registered; /* audio component registered in this driver */ struct drm_audio_component_audio_ops drm_audio_ops; + int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ struct hdac_chmap chmap; hda_nid_t vendor_nid; @@ -765,6 +769,10 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, static void jack_callback(struct hda_codec *codec, struct hda_jack_callback *jack) { + /* stop polling when notification is enabled */ + if (codec_has_acomp(codec)) + return; + /* hda_jack don't support DP MST */ check_presence_and_report(codec, jack->nid, 0); } @@ -823,6 +831,9 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + if (codec_has_acomp(codec)) + return; + if (!snd_hda_jack_tbl_get_from_tag(codec, tag)) { codec_dbg(codec, "Unexpected HDMI event tag 0x%x\n", tag); return; @@ -1421,7 +1432,7 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec, /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly */ -static void update_eld(struct hda_codec *codec, +static bool update_eld(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, struct hdmi_eld *eld) { @@ -1429,7 +1440,7 @@ static void update_eld(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; - int pcm_idx = -1; + int pcm_idx; /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; @@ -1452,18 +1463,22 @@ static void update_eld(struct hda_codec *codec, snd_hdmi_show_eld(codec, &eld->info); eld_changed = (pin_eld->eld_valid != eld->eld_valid); - if (eld->eld_valid && pin_eld->eld_valid) + eld_changed |= (pin_eld->monitor_present != eld->monitor_present); + if (!eld_changed && eld->eld_valid && pin_eld->eld_valid) if (pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, eld->eld_size) != 0) eld_changed = true; - pin_eld->monitor_present = eld->monitor_present; - pin_eld->eld_valid = eld->eld_valid; - pin_eld->eld_size = eld->eld_size; - if (eld->eld_valid) - memcpy(pin_eld->eld_buffer, eld->eld_buffer, eld->eld_size); - pin_eld->info = eld->info; + if (eld_changed) { + pin_eld->monitor_present = eld->monitor_present; + pin_eld->eld_valid = eld->eld_valid; + pin_eld->eld_size = eld->eld_size; + if (eld->eld_valid) + memcpy(pin_eld->eld_buffer, eld->eld_buffer, + eld->eld_size); + pin_eld->info = eld->info; + } /* * Re-setup pin and infoframe. This is needed e.g. when @@ -1481,6 +1496,7 @@ static void update_eld(struct hda_codec *codec, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); + return eld_changed; } /* update ELD and jack state via HD-audio verbs */ @@ -1582,6 +1598,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; struct snd_jack *jack = NULL; + bool changed; int size; mutex_lock(&per_pin->lock); @@ -1608,15 +1625,13 @@ static void sync_eld_via_acomp(struct hda_codec *codec, * disconnected event. Jack must be fetched before update_eld() */ jack = pin_idx_to_jack(codec, per_pin); - update_eld(codec, per_pin, eld); + changed = update_eld(codec, per_pin, eld); if (jack == NULL) jack = pin_idx_to_jack(codec, per_pin); - if (jack == NULL) - goto unlock; - snd_jack_report(jack, - (eld->monitor_present && eld->eld_valid) ? + if (changed && jack) + snd_jack_report(jack, + (eld->monitor_present && eld->eld_valid) ? SND_JACK_AVOUT : 0); - unlock: mutex_unlock(&per_pin->lock); } @@ -1632,18 +1647,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) snd_hda_power_down_pm(codec); return false; } - } - - if (codec_has_acomp(codec)) { + ret = hdmi_present_sense_via_verbs(per_pin, repoll); + snd_hda_power_down_pm(codec); + } else { sync_eld_via_acomp(codec, per_pin); ret = false; /* don't call snd_hda_jack_report_sync() */ - } else { - ret = hdmi_present_sense_via_verbs(per_pin, repoll); } - if (!codec_has_acomp(codec)) - snd_hda_power_down_pm(codec); - return ret; } @@ -2248,6 +2258,8 @@ static int generic_hdmi_init(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int pin_idx; + mutex_lock(&spec->pcm_lock); + spec->use_jack_detect = !codec->jackpoll_interval; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; @@ -2255,11 +2267,15 @@ static int generic_hdmi_init(struct hda_codec *codec) snd_hda_set_dev_select(codec, pin_nid, dev_id); hdmi_init_pin(codec, pin_nid); - if (!codec_has_acomp(codec)) + if (codec_has_acomp(codec)) + continue; + if (spec->use_jack_detect) + snd_hda_jack_detect_enable(codec, pin_nid); + else snd_hda_jack_detect_enable_callback(codec, pin_nid, - codec->jackpoll_interval > 0 ? - jack_callback : NULL); + jack_callback); } + mutex_unlock(&spec->pcm_lock); return 0; } @@ -2292,7 +2308,9 @@ static void generic_hdmi_free(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int pin_idx, pcm_idx; - if (codec_has_acomp(codec)) { + if (spec->acomp_registered) { + snd_hdac_acomp_exit(&codec->bus->core); + } else if (codec_has_acomp(codec)) { snd_hdac_acomp_register_notifier(&codec->bus->core, NULL); codec->relaxed_resume = 0; } @@ -2360,6 +2378,7 @@ static int alloc_generic_hdmi(struct hda_codec *codec) if (!spec) return -ENOMEM; + spec->codec = codec; spec->ops = generic_standard_hdmi_ops; spec->dev_num = 1; /* initialize to 1 */ mutex_init(&spec->pcm_lock); @@ -2398,6 +2417,138 @@ static int patch_generic_hdmi(struct hda_codec *codec) } /* + * generic audio component binding + */ + +/* turn on / off the unsol event jack detection dynamically */ +static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, + bool use_acomp) +{ + struct hda_jack_tbl *tbl; + + tbl = snd_hda_jack_tbl_get(codec, nid); + if (tbl) { + /* clear unsol even if component notifier is used, or re-enable + * if notifier is cleared + */ + unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, val); + } else { + /* if no jack entry was defined beforehand, create a new one + * at need (i.e. only when notifier is cleared) + */ + if (!use_acomp) + snd_hda_jack_detect_enable(codec, nid); + } +} + +/* set up / clear component notifier dynamically */ +static void generic_acomp_notifier_set(struct drm_audio_component *acomp, + bool use_acomp) +{ + struct hdmi_spec *spec; + int i; + + spec = container_of(acomp->audio_ops, struct hdmi_spec, drm_audio_ops); + mutex_lock(&spec->pcm_lock); + spec->use_acomp_notifier = use_acomp; + spec->codec->relaxed_resume = use_acomp; + /* reprogram each jack detection logic depending on the notifier */ + if (spec->use_jack_detect) { + for (i = 0; i < spec->num_pins; i++) + reprogram_jack_detect(spec->codec, + get_pin(spec, i)->pin_nid, + use_acomp); + } + mutex_unlock(&spec->pcm_lock); +} + +/* enable / disable the notifier via master bind / unbind */ +static int generic_acomp_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + generic_acomp_notifier_set(acomp, true); + return 0; +} + +static void generic_acomp_master_unbind(struct device *dev, + struct drm_audio_component *acomp) +{ + generic_acomp_notifier_set(acomp, false); +} + +/* check whether both HD-audio and DRM PCI devices belong to the same bus */ +static int match_bound_vga(struct device *dev, int subtype, void *data) +{ + struct hdac_bus *bus = data; + struct pci_dev *pci, *master; + + if (!dev_is_pci(dev) || !dev_is_pci(bus->dev)) + return 0; + master = to_pci_dev(bus->dev); + pci = to_pci_dev(dev); + return master->bus == pci->bus; +} + +/* audio component notifier for AMD/Nvidia HDMI codecs */ +static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) +{ + struct hda_codec *codec = audio_ptr; + struct hdmi_spec *spec = codec->spec; + hda_nid_t pin_nid = spec->port2pin(codec, port); + + if (!pin_nid) + return; + if (get_wcaps_type(get_wcaps(codec, pin_nid)) != AC_WID_PIN) + return; + /* skip notification during system suspend (but not in runtime PM); + * the state will be updated at resume + */ + if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + return; + /* ditto during suspend/resume process itself */ + if (snd_hdac_is_in_pm(&codec->core)) + return; + + check_presence_and_report(codec, pin_nid, dev_id); +} + +/* set up the private drm_audio_ops from the template */ +static void setup_drm_audio_ops(struct hda_codec *codec, + const struct drm_audio_component_audio_ops *ops) +{ + struct hdmi_spec *spec = codec->spec; + + spec->drm_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->drm_audio_ops.pin2port = ops->pin2port; + spec->drm_audio_ops.pin_eld_notify = ops->pin_eld_notify; + spec->drm_audio_ops.master_bind = ops->master_bind; + spec->drm_audio_ops.master_unbind = ops->master_unbind; +} + +/* initialize the generic HDMI audio component */ +static void generic_acomp_init(struct hda_codec *codec, + const struct drm_audio_component_audio_ops *ops, + int (*port2pin)(struct hda_codec *, int)) +{ + struct hdmi_spec *spec = codec->spec; + + spec->port2pin = port2pin; + setup_drm_audio_ops(codec, ops); + if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, + match_bound_vga, 0)) { + spec->acomp_registered = true; + codec->bus->keep_power = 0; + } +} + +/* * Intel codec parsers and helpers */ @@ -2565,20 +2716,19 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) check_presence_and_report(codec, pin_nid, dev_id); } +static const struct drm_audio_component_audio_ops intel_audio_ops = { + .pin2port = intel_pin2port, + .pin_eld_notify = intel_pin_eld_notify, +}; + /* register i915 component pin_eld_notify callback */ static void register_i915_notifier(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; spec->use_acomp_notifier = true; - spec->drm_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->drm_audio_ops.pin2port = intel_pin2port; - spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; + spec->port2pin = intel_port2pin; + setup_drm_audio_ops(codec, &intel_audio_ops); snd_hdac_acomp_register_notifier(&codec->bus->core, &spec->drm_audio_ops); /* no need for forcible resume for jack check thanks to notifier */ @@ -2612,6 +2762,8 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, /* precondition and allocation for Intel codecs */ static int alloc_intel_hdmi(struct hda_codec *codec) { + int err; + /* requires i915 binding */ if (!codec->bus->core.audio_component) { codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); @@ -2620,7 +2772,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec) return -ENODEV; } - return alloc_generic_hdmi(codec); + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + /* no need to handle unsol events */ + codec->patch_ops.unsol_event = NULL; + return 0; } /* parse and post-process for Intel codecs */ @@ -2976,6 +3133,7 @@ static int patch_simple_hdmi(struct hda_codec *codec, if (!spec) return -ENOMEM; + spec->codec = codec; codec->spec = spec; hdmi_array_init(spec, 1); @@ -3280,6 +3438,26 @@ static int nvhdmi_chmap_validate(struct hdac_chmap *chmap, return 0; } +/* map from pin NID to port; port is 0-based */ +/* for Nvidia: assume widget NID starting from 4, with step 1 (4, 5, 6, ...) */ +static int nvhdmi_pin2port(void *audio_ptr, int pin_nid) +{ + return pin_nid - 4; +} + +/* reverse-map from port to pin NID: see above */ +static int nvhdmi_port2pin(struct hda_codec *codec, int port) +{ + return port + 4; +} + +static const struct drm_audio_component_audio_ops nvhdmi_audio_ops = { + .pin2port = nvhdmi_pin2port, + .pin_eld_notify = generic_acomp_pin_eld_notify, + .master_bind = generic_acomp_master_bind, + .master_unbind = generic_acomp_master_unbind, +}; + static int patch_nvhdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3296,6 +3474,10 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); + return 0; } @@ -3783,6 +3965,26 @@ static int atihdmi_init(struct hda_codec *codec) return 0; } +/* map from pin NID to port; port is 0-based */ +/* for AMD: assume widget NID starting from 3, with step 2 (3, 5, 7, ...) */ +static int atihdmi_pin2port(void *audio_ptr, int pin_nid) +{ + return pin_nid / 2 - 1; +} + +/* reverse-map from port to pin NID: see above */ +static int atihdmi_port2pin(struct hda_codec *codec, int port) +{ + return port * 2 + 3; +} + +static const struct drm_audio_component_audio_ops atihdmi_audio_ops = { + .pin2port = atihdmi_pin2port, + .pin_eld_notify = generic_acomp_pin_eld_notify, + .master_bind = generic_acomp_master_bind, + .master_unbind = generic_acomp_master_unbind, +}; + static int patch_atihdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3831,6 +4033,8 @@ static int patch_atihdmi(struct hda_codec *codec) */ codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &atihdmi_audio_ops, atihdmi_port2pin); + return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de224cbea7a0..085a2f95e076 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -393,6 +393,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: alc_update_coef_idx(codec, 0x10, 1<<15, 0); break; case 0x10ec0662: @@ -837,9 +838,11 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + spec->gen.skip_verbs = 1; /* applied in below */ snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); + snd_hda_apply_verbs(codec); /* apply verbs here after own init */ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); @@ -869,15 +872,6 @@ static void alc_reboot_notify(struct hda_codec *codec) alc_shutup(codec); } -/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */ -static void alc_d3_at_reboot(struct hda_codec *codec) -{ - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); -} - #define alc_free snd_hda_gen_free #ifdef CONFIG_PM @@ -1065,6 +1059,9 @@ static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), + /* blacklist -- no beep available */ + SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0), + SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0), {} }; @@ -2848,7 +2845,8 @@ static int patch_alc268(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x01; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; spec->shutup = alc_eapd_shutup; @@ -3762,6 +3760,72 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec, vref); } +/* + * Magic sequence to make Huawei Matebook X right speaker working (bko#197801) + */ +struct hda_alc298_mbxinit { + unsigned char value_0x23; + unsigned char value_0x25; +}; + +static void alc298_huawei_mbx_stereo_seq(struct hda_codec *codec, + const struct hda_alc298_mbxinit *initval, + bool first) +{ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x0); + alc_write_coef_idx(codec, 0x26, 0xb000); + + if (first) + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_GET_PIN_SENSE, 0x0); + + snd_hda_codec_write(codec, 0x6, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x23, initval->value_0x23); + + if (initval->value_0x23 != 0x1e) + alc_write_coef_idx(codec, 0x25, initval->value_0x25); + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); +} + +static void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* Initialization magic */ + static const struct hda_alc298_mbxinit dac_init[] = { + {0x0c, 0x00}, {0x0d, 0x00}, {0x0e, 0x00}, {0x0f, 0x00}, + {0x10, 0x00}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x00}, + {0x1d, 0x00}, {0x1e, 0x00}, {0x1f, 0x00}, + {0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24}, + {0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f}, + {0x2b, 0x02}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x00}, + {0x2f, 0x00}, + {0x30, 0x00}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0x00}, + {0x34, 0x00}, {0x35, 0x01}, {0x36, 0x93}, {0x37, 0x0c}, + {0x38, 0x00}, {0x39, 0x00}, {0x3a, 0xf8}, {0x38, 0x80}, + {} + }; + const struct hda_alc298_mbxinit *seq; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + /* Start */ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x00); + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x22, 0x31); + alc_write_coef_idx(codec, 0x23, 0x0b); + alc_write_coef_idx(codec, 0x25, 0x00); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); + + for (seq = dac_init; seq->value_0x23; seq++) + alc298_huawei_mbx_stereo_seq(codec, seq, seq == dac_init); +} + static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5152,7 +5216,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ + spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); @@ -5295,6 +5359,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5754,10 +5829,12 @@ enum { ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, + ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -5787,6 +5864,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MACH_WX9_PINS, + ALC298_FIXUP_HUAWEI_MBX_STEREO, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -5804,8 +5882,11 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, + ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, + ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6095,6 +6176,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, + [ALC298_FIXUP_HUAWEI_MBX_STEREO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_huawei_mbx_stereo, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6435,6 +6522,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* headset mic w/o jack detect */ + { } + }, + .chained_before = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6476,6 +6572,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -6830,6 +6932,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6846,6 +6957,26 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, + [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x411111f0 }, /* disable confusing internal speaker */ + { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11040 }, + { 0x21, 0x04211020 }, + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6899,17 +7030,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), @@ -6987,6 +7118,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7003,6 +7136,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), @@ -7080,6 +7215,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -7105,6 +7241,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -7272,6 +7409,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, + {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {} }; #define ALC225_STANDARD_PINS \ @@ -7582,10 +7721,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, - {0x12, 0xb7a60130}, - {0x14, 0x90170110}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x15, 0x04211040}, @@ -7688,6 +7823,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x90170110}, {0x1a, 0x03011020}, {0x21, 0x03211030}), + SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, + {0x12, 0xb7a60140}, + {0x17, 0x90170110}, + {0x1a, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, @@ -7695,6 +7835,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {} }; +/* This is the fallback pin_fixup_tbl for alc269 family, to make the tbl match + * more machines, don't need to match all valid pins, just need to match + * all the pins defined in the tbl. Just because of this reason, it is possible + * that a single machine matches multiple tbls, so there is one limitation: + * at most one tbl is allowed to define for the same vendor and same codec + */ +static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + {0x19, 0x40000000}, + {0x1b, 0x40000000}), + {} +}; + static void alc269_fill_coef(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -7867,6 +8020,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ @@ -7884,7 +8038,8 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); + snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -8018,7 +8173,8 @@ static int patch_alc861(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x23; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x23; #ifdef CONFIG_PM spec->power_hook = alc_power_eapd; @@ -8119,7 +8275,8 @@ static int patch_alc861vd(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x23; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x23; spec->shutup = alc_eapd_shutup; @@ -8259,6 +8416,45 @@ static void alc662_fixup_usi_headset_mic(struct hda_codec *codec, } } +static void alc662_aspire_ethos_mute_speakers(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* surround speakers at 0x1b already get muted automatically when + * headphones are plugged in, but we have to mute/unmute the remaining + * channels manually: + * 0x15 - front left/front right + * 0x18 - front center/ LFE + */ + if (snd_hda_jack_detect_state(codec, 0x1b) == HDA_JACK_PRESENT) { + snd_hda_set_pin_ctl_cache(codec, 0x15, 0); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } else { + snd_hda_set_pin_ctl_cache(codec, 0x15, PIN_OUT); + snd_hda_set_pin_ctl_cache(codec, 0x18, PIN_OUT); + } +} + +static void alc662_fixup_aspire_ethos_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x1b: shared headphones jack and surround speakers */ + if (!is_jack_detectable(codec, 0x1b)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x1b, + alc662_aspire_ethos_mute_speakers); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc662_aspire_ethos_mute_speakers(codec, NULL); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -8330,6 +8526,9 @@ enum { ALC662_FIXUP_USI_FUNC, ALC662_FIXUP_USI_HEADSET_MODE, ALC662_FIXUP_LENOVO_MULTI_CODECS, + ALC669_FIXUP_ACER_ASPIRE_ETHOS, + ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER, + ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET, }; static const struct hda_fixup alc662_fixups[] = { @@ -8656,6 +8855,33 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_alc662_fixup_lenovo_dual_codecs, }, + [ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_aspire_ethos_hp, + }, + [ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER] = { + .type = HDA_FIXUP_VERBS, + /* subwoofer needs an extra GPIO setting to become audible */ + .v.verbs = (const struct hda_verb[]) { + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + { } + }, + .chained = true, + .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET + }, + [ALC669_FIXUP_ACER_ASPIRE_ETHOS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x15, 0x92130110 }, /* front speakers */ + { 0x18, 0x99130111 }, /* center/subwoofer */ + { 0x1b, 0x11130012 }, /* surround plus jack for HP */ + { } + }, + .chained = true, + .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -8701,6 +8927,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), + SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), #if 0 /* Below is a quirk table taken from the old code. @@ -8794,6 +9021,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"}, {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, + {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, {} }; @@ -8869,7 +9097,7 @@ static int patch_alc662(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); - snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups); + snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -8954,6 +9182,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), @@ -9006,6 +9235,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269), HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0d9b62768241..894f3f509e76 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -975,15 +975,6 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec) return 0; } -/* - */ - -static const struct hda_verb stac9200_core_init[] = { - /* set dac0mux for dac converter */ - { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - static const struct hda_verb stac9200_eapd_init[] = { /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 583ca7384d83..fe10714380f2 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -49,6 +49,14 @@ static const struct pci_device_id snd_lx6464es_ids[] = { PCI_VENDOR_ID_DIGIGRAM, PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM), }, /* LX6464ES-CAE */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe-CAE */ { 0, }, }; diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 9ca9214cb7fb..5f40517717c4 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -10,7 +10,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR - depends on SND_SOC_AMD_ACP && I2C + depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 9a406144b18f..befc2a3a05b0 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -674,8 +674,13 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, dev->channels = channels; ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra); - if (ret < 0) + if (ret < 0) { + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } return ret; + } return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb); } @@ -690,31 +695,37 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream, err = wait_event_interruptible_timeout(dev->wq_txrdy, dev->tx_rdy, msecs_to_jiffies(500)); + if (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Tx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels)); + dev->tx_rdy = 1; + } } else { err = wait_event_interruptible_timeout(dev->wq_rxrdy, dev->rx_rdy, msecs_to_jiffies(500)); - } - - if (err == 0) { - u32 idra; - - dev_warn_once(dev->dev, "Timeout waiting for %s\n", - is_playback ? "Tx ready" : "Rx ready"); - if (is_playback) - idra = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels); - else - idra = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels); - regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra); + if (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Rx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)); + dev->rx_rdy = 1; + } } if (!mchp_i2s_mcc_is_running(dev)) { regmap_write(dev->regmap, MCHP_I2SMCC_CR, MCHP_I2SMCC_CR_CKDIS); if (dev->gclk_running) { - clk_disable_unprepare(dev->gclk); + clk_disable(dev->gclk); dev->gclk_running = 0; } + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } } return 0; @@ -813,6 +824,8 @@ static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai) init_waitqueue_head(&dev->wq_txrdy); init_waitqueue_head(&dev->wq_rxrdy); + dev->tx_rdy = 1; + dev->rx_rdy = 1; snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 89238343e34d..229cc89f8c5a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,7 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_BD28623 select SND_SOC_CQ0093VC - select SND_SOC_CROS_EC_CODEC if MFD_CROS_EC + select SND_SOC_CROS_EC_CODEC if CROS_EC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C @@ -477,7 +477,7 @@ config SND_SOC_CQ0093VC config SND_SOC_CROS_EC_CODEC tristate "codec driver for ChromeOS EC" - depends on MFD_CROS_EC + depends on CROS_EC help If you say yes here you will get support for the ChromeOS Embedded Controller's Audio Codec. diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 85beef265cc8..3c1bd24a1057 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -9,9 +9,9 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/kernel.h> -#include <linux/mfd/cros_ec.h> -#include <linux/mfd/cros_ec_commands.h> #include <linux/module.h> +#include <linux/platform_data/cros_ec_commands.h> +#include <linux/platform_data/cros_ec_proto.h> #include <linux/platform_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 9150e7068467..36eef1fb3d18 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -53,7 +53,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -91,7 +94,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 4, 0, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1628,14 +1628,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream, static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int reg_val = 0, reg_bclk_inv = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: reg_val |= RT1011_I2S_TDM_MS_S; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1645,7 +1649,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_bclk_inv |= RT1011_TDM_INV_BCLK; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1661,7 +1665,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_val |= RT1011_I2S_TDM_DF_PCM_B; break; default: - return -EINVAL; + ret = -EINVAL; } switch (dai->id) { @@ -1676,9 +1680,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); - return -EINVAL; + ret = -EINVAL; } - return 0; + + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_set_component_sysclk(struct snd_soc_component *component, @@ -1797,8 +1803,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int val = 0, tdm_en = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) tdm_en = RT1011_TDM_I2S_DOCK_EN_1; @@ -1818,7 +1828,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 2: break; default: - return -EINVAL; + ret = -EINVAL; } switch (slot_width) { @@ -1837,7 +1847,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 16: break; default: - return -EINVAL; + ret = -EINVAL; } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1854,7 +1864,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); - return 0; + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_probe(struct snd_soc_component *component) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 54ac2fd41925..67f06c95eec5 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -6,6 +6,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index eaf3e2208a06..70bb86f3342f 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -12,6 +12,7 @@ */ #include <linux/dmi.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 4977b5a65eb8..9d657421730a 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -8,6 +8,7 @@ * Mengdong Lin <mengdong.lin@intel.com> */ +#include <linux/gpio/consumer.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index e6357d306cb8..863a477d3405 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -36,7 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT); SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT); SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT); SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS); -SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); +SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 1186a03a88d6..6068bb697e22 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -223,6 +223,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, if (ipc->ops.reply_msg_match != NULL) header = ipc->ops.reply_msg_match(header, &mask); + else + mask = (u64)-1; if (list_empty(&ipc->rx_list)) { dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index 212370bf704c..3466675f2678 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -188,7 +188,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, memset(d->fw_read_buff, 0, FW_REG_BUF); if (w0_stat_sz > 0) - __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); + __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); for (offset = 0; offset < FW_REG_SIZE; offset += 16) { ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index ab3d23c7bd65..19f328d71f24 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -136,7 +136,7 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; - dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", + dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", nhlt->header.oem_id, nhlt->header.oem_table_id, nhlt->header.oem_revision); diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 6b757168693e..e72f826062e9 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,30 +1,9 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_JZ4740_SOC - tristate "SoC Audio for Ingenic JZ4740 SoC" - depends on MIPS || COMPILE_TEST - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to - the JZ4740 I2S interface. You will also need to select the audio - interfaces to support below. - -if SND_JZ4740_SOC - config SND_JZ4740_SOC_I2S tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC" + depends on MIPS || COMPILE_TEST depends on HAS_IOMEM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740 based boards. - -config SND_JZ4740_SOC_QI_LB60 - tristate "SoC Audio support for Qi LB60" - depends on HAS_IOMEM - depends on JZ4740_QI_LB60 || COMPILE_TEST - select SND_JZ4740_SOC_I2S - select SND_SOC_JZ4740_CODEC - help - Say Y if you want to add support for ASoC audio on the Qi LB60 board - a.k.a Qi Ben NanoNote. - -endif diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index fb10e9ad9ff7..f8701c9b09fe 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -5,8 +5,3 @@ snd-soc-jz4740-i2s-objs := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o - -# Jz4740 Machine Support -snd-soc-qi-lb60-objs := qi_lb60.o - -obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c deleted file mode 100644 index 8ef6f41dcfbe..000000000000 --- a/sound/soc/jz4740/qi_lb60.c +++ /dev/null @@ -1,106 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <linux/gpio/consumer.h> - -struct qi_lb60 { - struct gpio_desc *snd_gpio; - struct gpio_desc *amp_gpio; -}; - -static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *ctrl, int event) -{ - struct qi_lb60 *qi_lb60 = snd_soc_card_get_drvdata(widget->dapm->card); - int on = !SND_SOC_DAPM_EVENT_OFF(event); - - gpiod_set_value_cansleep(qi_lb60->snd_gpio, on); - gpiod_set_value_cansleep(qi_lb60->amp_gpio, on); - - return 0; -} - -static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route qi_lb60_routes[] = { - {"Mic", NULL, "MIC"}, - {"Speaker", NULL, "LOUT"}, - {"Speaker", NULL, "ROUT"}, -}; - -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_CPU("jz4740-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("jz4740-codec", "jz4740-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("jz4740-i2s"))); - -static struct snd_soc_dai_link qi_lb60_dai = { - .name = "jz4740", - .stream_name = "jz4740", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - SND_SOC_DAILINK_REG(hifi), -}; - -static struct snd_soc_card qi_lb60_card = { - .name = "QI LB60", - .owner = THIS_MODULE, - .dai_link = &qi_lb60_dai, - .num_links = 1, - - .dapm_widgets = qi_lb60_widgets, - .num_dapm_widgets = ARRAY_SIZE(qi_lb60_widgets), - .dapm_routes = qi_lb60_routes, - .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), - .fully_routed = true, -}; - -static int qi_lb60_probe(struct platform_device *pdev) -{ - struct qi_lb60 *qi_lb60; - struct snd_soc_card *card = &qi_lb60_card; - - qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL); - if (!qi_lb60) - return -ENOMEM; - - qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW); - if (IS_ERR(qi_lb60->snd_gpio)) - return PTR_ERR(qi_lb60->snd_gpio); - - qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW); - if (IS_ERR(qi_lb60->amp_gpio)) - return PTR_ERR(qi_lb60->amp_gpio); - - card->dev = &pdev->dev; - - snd_soc_card_set_drvdata(card, qi_lb60); - - return devm_snd_soc_register_card(&pdev->dev, card); -} - -static struct platform_driver qi_lb60_driver = { - .driver = { - .name = "qi-lb60-audio", - }, - .probe = qi_lb60_probe, -}; - -module_platform_driver(qi_lb60_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:qi-lb60-audio"); diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index d16563408465..10ea4fdbeb1e 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; int hd_audio = 0; - int hd_align = 1; + int hd_align = 0; /* set hd mode */ switch (substream->runtime->format) { @@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S24_LE: hd_audio = 1; - hd_align = 0; break; default: dev_err(afe->dev, "%s() error: unsupported format %d\n", diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8e3e86619b35..60086858e920 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,7 +99,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && MFD_CROS_EC && I2C + depends on QCOM_APR && CROS_EC && I2C select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index c16b0ffe8cfc..d951100bf770 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -422,11 +422,6 @@ static const struct dailink_match_data dailink_match[] = { }, }; -static int of_dev_node_match(struct device *dev, const void *data) -{ - return dev->of_node == data; -} - static int rockchip_sound_codec_node_match(struct device_node *np_codec) { struct device *dev; @@ -438,8 +433,8 @@ static int rockchip_sound_codec_node_match(struct device_node *np_codec) continue; if (dailink_match[i].bus_type) { - dev = bus_find_device(dailink_match[i].bus_type, NULL, - np_codec, of_dev_node_match); + dev = bus_find_device_by_of_node(dailink_match[i].bus_type, + np_codec); if (!dev) continue; put_device(dev); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..5552c66ca642 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -306,6 +306,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strscpy_pad(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c25939c5611b..0fd032914a31 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -80,12 +80,6 @@ struct soc_tplg { static int soc_tplg_process_headers(struct soc_tplg *tplg); static void soc_tplg_complete(struct soc_tplg *tplg); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); static void soc_tplg_denum_remove_texts(struct soc_enum *se); static void soc_tplg_denum_remove_values(struct soc_enum *se); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index f04d9fb5130f..d89b5c928c4d 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -187,57 +187,9 @@ static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); u32 spcr; u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - if (spcr & mask) { - /* start off disabled */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - spcr & ~mask); - toggle_clock(dev, playback); - } - if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { - /* Start the sample generator */ - spcr |= DAVINCI_MCBSP_SPCR_GRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - } - - if (playback) { - /* Stop the DMA to avoid data loss */ - /* while the transmitter is out of reset to handle XSYNCERR */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA stop failed\n"); - } - - /* Enable the transmitter */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr |= DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - - /* wait for any unexpected frame sync error to occur */ - udelay(100); - - /* Disable the transmitter to clear any outstanding XSYNCERR */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr &= ~DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - toggle_clock(dev, playback); - - /* Restart the DMA */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA start failed\n"); - } - } /* Enable transmitter or receiver */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -575,7 +527,41 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 spcr; + u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; + davinci_mcbsp_stop(dev, playback); + + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (spcr & mask) { + /* start off disabled */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + spcr & ~mask); + toggle_clock(dev, playback); + } + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { + /* Start the sample generator */ + spcr |= DAVINCI_MCBSP_SPCR_GRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } + + if (playback) { + /* Enable the transmitter */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); + } + return 0; } diff --git a/sound/sound_core.c b/sound/sound_core.c index b730d97c4de6..90d118cd9164 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -275,7 +275,8 @@ retry: goto retry; } spin_unlock(&sound_loader_lock); - return -EBUSY; + r = -EBUSY; + goto fail; } } diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 010113156239..6e065d44060e 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -580,12 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len) switch (len) { case 32: b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); + /* fall through */ case 16: b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); + /* fall through */ case 8: b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); + /* fall through */ case 4: b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); + /* fall through */ case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); case 1: diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e1ce257ab705..78edd7d2f418 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,12 +11,14 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ proc.o \ quirks.o \ - stream.o + stream.o \ + validate.o snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 72e9bdf76115..6b8c14f9b5d4 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -38,39 +38,37 @@ static void *find_uac_clock_desc(struct usb_host_interface *iface, int id, static bool validate_clock_source_v2(void *p, int id) { struct uac_clock_source_descriptor *cs = p; - return cs->bLength == sizeof(*cs) && cs->bClockID == id; + return cs->bClockID == id; } static bool validate_clock_source_v3(void *p, int id) { struct uac3_clock_source_descriptor *cs = p; - return cs->bLength == sizeof(*cs) && cs->bClockID == id; + return cs->bClockID == id; } static bool validate_clock_selector_v2(void *p, int id) { struct uac_clock_selector_descriptor *cs = p; - return cs->bLength >= sizeof(*cs) && cs->bClockID == id && - cs->bLength == 7 + cs->bNrInPins; + return cs->bClockID == id; } static bool validate_clock_selector_v3(void *p, int id) { struct uac3_clock_selector_descriptor *cs = p; - return cs->bLength >= sizeof(*cs) && cs->bClockID == id && - cs->bLength == 11 + cs->bNrInPins; + return cs->bClockID == id; } static bool validate_clock_multiplier_v2(void *p, int id) { struct uac_clock_multiplier_descriptor *cs = p; - return cs->bLength == sizeof(*cs) && cs->bClockID == id; + return cs->bClockID == id; } static bool validate_clock_multiplier_v3(void *p, int id) { struct uac3_clock_multiplier_descriptor *cs = p; - return cs->bLength == sizeof(*cs) && cs->bClockID == id; + return cs->bClockID == id; } #define DEFINE_FIND_HELPER(name, obj, validator, type) \ diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 71d5f540334a..4c12cc5b53fd 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -72,7 +72,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) struct usb_host_endpoint *ep; ep = usb_pipe_endpoint(dev, pipe); - if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) return -EINVAL; return 0; } diff --git a/sound/usb/helper.h b/sound/usb/helper.h index 6afb70156ec4..5e8a18b4e7b9 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -31,4 +31,8 @@ static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip) return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber; } +/* in validate.c */ +bool snd_usb_validate_audio_desc(void *p, int protocol); +bool snd_usb_validate_midi_desc(void *p); + #endif /* __USBAUDIO_HELPER_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1d5d13..c406497c5919 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); if (ret < 0) - return ret; + goto error; } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); if (ret < 0) { - kfree(rt); dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); - return ret; + goto error; } pcm->private_data = rt; @@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) chip->pcm = rt; return 0; + +error: + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + kfree(rt); + return ret; } diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index ab2ec896f49c..b5a3f754a4f1 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -342,7 +342,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, if (address > 0xffff || datalen > 0xff) return -EINVAL; - len = kmalloc(sizeof(*len), GFP_KERNEL); + len = kmalloc(1, GFP_KERNEL); if (!len) return -ENOMEM; @@ -418,7 +418,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, if (address > 0xffff || datalen > 0xffff) return -EINVAL; - status = kmalloc(sizeof(*status), GFP_KERNEL); + status = kmalloc(1, GFP_KERNEL); if (!status) return -ENOMEM; diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 2c03e0f6bf72..f70211e6b174 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->volume_monitor = 255; line6pcm->line6 = line6; + spin_lock_init(&line6pcm->out.lock); + spin_lock_init(&line6pcm->in.lock); + line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; + + line6->line6pcm = line6pcm; + + pcm->private_data = line6pcm; + pcm->private_free = line6_cleanup_pcm; + line6pcm->max_packet_size_in = usb_maxpacket(line6->usbdev, usb_rcvisocpipe(line6->usbdev, ep_read), 0); @@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6, return -EINVAL; } - spin_lock_init(&line6pcm->out.lock); - spin_lock_init(&line6pcm->in.lock); - line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; - - line6->line6pcm = line6pcm; - - pcm->private_data = line6pcm; - pcm->private_free = line6_cleanup_pcm; - err = line6_create_audio_out_urbs(line6pcm); if (err < 0) return err; diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index f0662bd4e50f..27bf61c177c0 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = { .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_HWMON, - .altsetting = 1, + .altsetting = 0, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0d24c72c155f..ed158f04de80 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -244,5 +244,5 @@ static struct usb_driver variax_driver = { module_usb_driver(variax_driver); -MODULE_DESCRIPTION("Vairax Workbench USB driver"); +MODULE_DESCRIPTION("Variax Workbench USB driver"); MODULE_LICENSE("GPL"); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7498b5191b68..3fd1d1749edf 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -68,6 +68,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -738,12 +739,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; - void *c; - - if (desc->bLength < sizeof(*desc)) - return -EINVAL; - if (!desc->bNrInPins) - return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: @@ -759,229 +754,260 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, break; } - if (!mu_channels) - return 0; - - c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); - if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) - return 0; /* no bmControls -> skip */ - return mu_channels; } /* - * parse the source unit recursively until it reaches to a terminal - * or a branched unit. + * Parse Input Terminal Unit */ -static int check_input_term(struct mixer_build *state, int id, - struct usb_audio_term *term) +static int __check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term); + +static int parse_term_uac1_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) { - int protocol = state->mixer->protocol; + struct uac_input_terminal_descriptor *d = p1; + + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le16_to_cpu(d->wChannelConfig); + term->name = d->iTerminal; + return 0; +} + +static int parse_term_uac2_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac2_input_terminal_descriptor *d = p1; int err; - void *p1; - memset(term, 0, sizeof(*term)); - while ((p1 = find_audio_control_unit(state, id)) != NULL) { - unsigned char *hdr = p1; - term->id = id; + /* call recursively to verify the referenced clock entity */ + err = __check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; - if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { - switch (hdr[2]) { - case UAC_INPUT_TERMINAL: - if (protocol == UAC_VERSION_1) { - struct uac_input_terminal_descriptor *d = p1; - - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le16_to_cpu(d->wChannelConfig); - term->name = d->iTerminal; - } else { /* UAC_VERSION_2 */ - struct uac2_input_terminal_descriptor *d = p1; - - /* call recursively to verify that the - * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); - if (err < 0) - return err; + /* save input term properties after recursion, + * to ensure they are not overriden by the recursion calls + */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; + return 0; +} - /* save input term properties after recursion, - * to ensure they are not overriden by the - * recursion calls */ - term->id = id; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - } - return 0; - case UAC_FEATURE_UNIT: { - /* the header is the same for v1 and v2 */ - struct uac_feature_unit_descriptor *d = p1; +static int parse_term_uac3_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac3_input_terminal_descriptor *d = p1; + int err; - id = d->bSourceID; - break; /* continue to parse */ - } - case UAC_MIXER_UNIT: { - struct uac_mixer_unit_descriptor *d = p1; - - term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ - term->channels = uac_mixer_unit_bNrChannels(d); - term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); - term->name = uac_mixer_unit_iMixer(d); - return 0; - } - case UAC_SELECTOR_UNIT: - case UAC2_CLOCK_SELECTOR: { - struct uac_selector_unit_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; - term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ - term->id = id; - term->name = uac_selector_unit_iSelector(d); - return 0; - } - case UAC1_PROCESSING_UNIT: - /* UAC2_EFFECT_UNIT */ - if (protocol == UAC_VERSION_1) - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - else /* UAC_VERSION_2 */ - term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ - /* fall through */ - case UAC1_EXTENSION_UNIT: - /* UAC2_PROCESSING_UNIT_V2 */ - if (protocol == UAC_VERSION_1 && !term->type) - term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ - else if (protocol == UAC_VERSION_2 && !term->type) - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - /* fall through */ - case UAC2_EXTENSION_UNIT_V2: { - struct uac_processing_unit_descriptor *d = p1; - - if (protocol == UAC_VERSION_2 && - hdr[2] == UAC2_EFFECT_UNIT) { - /* UAC2/UAC1 unit IDs overlap here in an - * uncompatible way. Ignore this unit for now. - */ - return 0; - } + /* call recursively to verify the referenced clock entity */ + err = __check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; - if (d->bNrInPins) { - id = d->baSourceID[0]; - break; /* continue to parse */ - } - if (!term->type) - term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + /* save input term properties after recursion, + * to ensure they are not overriden by the recursion calls + */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); - term->channels = uac_processing_unit_bNrChannels(d); - term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); - term->name = uac_processing_unit_iProcessing(d, protocol); - return 0; - } - case UAC2_CLOCK_SOURCE: { - struct uac_clock_source_descriptor *d = p1; + err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); + if (err < 0) + return err; + term->channels = err; - term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ - term->id = id; - term->name = d->iClockSource; - return 0; - } - default: - return -ENODEV; - } - } else { /* UAC_VERSION_3 */ - switch (hdr[2]) { - case UAC_INPUT_TERMINAL: { - struct uac3_input_terminal_descriptor *d = p1; - - /* call recursively to verify that the - * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); - if (err < 0) - return err; + /* REVISIT: UAC3 IT doesn't have channels cfg */ + term->chconfig = 0; - /* save input term properties after recursion, - * to ensure they are not overriden by the - * recursion calls */ - term->id = id; - term->type = le16_to_cpu(d->wTerminalType); + term->name = le16_to_cpu(d->wTerminalDescrStr); + return 0; +} - err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); - if (err < 0) - return err; - term->channels = err; +static int parse_term_mixer_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_mixer_unit_descriptor *d = p1; + int protocol = state->mixer->protocol; + int err; - /* REVISIT: UAC3 IT doesn't have channels cfg */ - term->chconfig = 0; + err = uac_mixer_unit_get_channels(state, d); + if (err <= 0) + return err; - term->name = le16_to_cpu(d->wTerminalDescrStr); - return 0; - } - case UAC3_FEATURE_UNIT: { - struct uac3_feature_unit_descriptor *d = p1; + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ + term->channels = err; + if (protocol != UAC_VERSION_3) { + term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); + term->name = uac_mixer_unit_iMixer(d); + } + return 0; +} - id = d->bSourceID; - break; /* continue to parse */ - } - case UAC3_CLOCK_SOURCE: { - struct uac3_clock_source_descriptor *d = p1; +static int parse_term_selector_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_selector_unit_descriptor *d = p1; + int err; - term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ - term->id = id; - term->name = le16_to_cpu(d->wClockSourceStr); - return 0; - } - case UAC3_MIXER_UNIT: { - struct uac_mixer_unit_descriptor *d = p1; + /* call recursively to retrieve the channel info */ + err = __check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ + term->id = id; + if (state->mixer->protocol != UAC_VERSION_3) + term->name = uac_selector_unit_iSelector(d); + return 0; +} - err = uac_mixer_unit_get_channels(state, d); - if (err <= 0) - return err; +static int parse_term_proc_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id, int vtype) +{ + struct uac_processing_unit_descriptor *d = p1; + int protocol = state->mixer->protocol; + int err; - term->channels = err; - term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ + if (d->bNrInPins) { + /* call recursively to retrieve the channel info */ + err = __check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + } - return 0; - } - case UAC3_SELECTOR_UNIT: - case UAC3_CLOCK_SELECTOR: { - struct uac_selector_unit_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; - term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ - term->id = id; - term->name = 0; /* TODO: UAC3 Class-specific strings */ + term->type = vtype << 16; /* virtual type */ + term->id = id; - return 0; - } - case UAC3_PROCESSING_UNIT: { - struct uac_processing_unit_descriptor *d = p1; + if (protocol == UAC_VERSION_3) + return 0; - if (!d->bNrInPins) - return -EINVAL; + if (!term->channels) { + term->channels = uac_processing_unit_bNrChannels(d); + term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); + } + term->name = uac_processing_unit_iProcessing(d, protocol); + return 0; +} - /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; +static int parse_term_uac2_clock_source(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_clock_source_descriptor *d = p1; - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - term->id = id; - term->name = 0; /* TODO: UAC3 Class-specific strings */ + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ + term->id = id; + term->name = d->iClockSource; + return 0; +} - return 0; - } - default: - return -ENODEV; - } +static int parse_term_uac3_clock_source(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac3_clock_source_descriptor *d = p1; + + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ + term->id = id; + term->name = le16_to_cpu(d->wClockSourceStr); + return 0; +} + +#define PTYPE(a, b) ((a) << 8 | (b)) + +/* + * parse the source unit recursively until it reaches to a terminal + * or a branched unit. + */ +static int __check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + int protocol = state->mixer->protocol; + void *p1; + unsigned char *hdr; + + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + if (!snd_usb_validate_audio_desc(p1, protocol)) + break; /* bad descriptor */ + + hdr = p1; + term->id = id; + + switch (PTYPE(protocol, hdr[2])) { + case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT): { + /* the header is the same for all versions */ + struct uac_feature_unit_descriptor *d = p1; + + id = d->bSourceID; + break; /* continue to parse */ + } + case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL): + return parse_term_uac1_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL): + return parse_term_uac2_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL): + return parse_term_uac3_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT): + return parse_term_mixer_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR): + case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR): + return parse_term_selector_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_PROCESSING_UNIT); + case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_EFFECT_UNIT); + case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_EXTENSION_UNIT); + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE): + return parse_term_uac2_clock_source(state, term, p1, id); + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE): + return parse_term_uac3_clock_source(state, term, p1, id); + default: + return -ENODEV; } } return -ENODEV; } + +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + memset(term, 0, sizeof(*term)); + memset(state->termbitmap, 0, sizeof(state->termbitmap)); + return __check_input_term(state, id, term); +} + /* * Feature Unit */ @@ -1011,10 +1037,15 @@ static struct usb_feature_control_info audio_feature_info[] = { { UAC2_FU_PHASE_INVERTER, "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; +static void usb_mixer_elem_info_free(struct usb_mixer_elem_info *cval) +{ + kfree(cval); +} + /* private_free callback */ void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) { - kfree(kctl->private_data); + usb_mixer_elem_info_free(kctl->private_data); kctl->private_data = NULL; } @@ -1537,7 +1568,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, ctl_info = get_feature_control_info(control); if (!ctl_info) { - kfree(cval); + usb_mixer_elem_info_free(cval); return; } if (mixer->protocol == UAC_VERSION_1) @@ -1570,7 +1601,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, if (!kctl) { usb_audio_err(mixer->chip, "cannot malloc kcontrol\n"); - kfree(cval); + usb_mixer_elem_info_free(cval); return; } kctl->private_free = snd_usb_mixer_elem_free; @@ -1740,7 +1771,7 @@ static void build_connector_control(struct usb_mixer_interface *mixer, kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval); if (!kctl) { usb_audio_err(mixer->chip, "cannot malloc kcontrol\n"); - kfree(cval); + usb_mixer_elem_info_free(cval); return; } get_connector_control_name(mixer, term, is_input, kctl->id.name, @@ -1761,13 +1792,6 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, if (state->mixer->protocol != UAC_VERSION_2) return -EINVAL; - if (hdr->bLength != sizeof(*hdr)) { - usb_audio_dbg(state->chip, - "Bogus clock source descriptor length of %d, ignoring.\n", - hdr->bLength); - return 0; - } - /* * The only property of this unit we are interested in is the * clock source validity. If that isn't readable, just bail out. @@ -1793,7 +1817,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval); if (!kctl) { - kfree(cval); + usb_mixer_elem_info_free(cval); return -ENOMEM; } @@ -1826,62 +1850,20 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, __u8 *bmaControls; if (state->mixer->protocol == UAC_VERSION_1) { - if (hdr->bLength < 7) { - usb_audio_err(state->chip, - "unit %u: invalid UAC_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } csize = hdr->bControlSize; - if (!csize) { - usb_audio_dbg(state->chip, - "unit %u: invalid bControlSize == 0\n", - unitid); - return -EINVAL; - } channels = (hdr->bLength - 7) / csize - 1; bmaControls = hdr->bmaControls; - if (hdr->bLength < 7 + csize) { - usb_audio_err(state->chip, - "unit %u: invalid UAC_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } } else if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_feature_unit_descriptor *ftr = _ftr; - if (hdr->bLength < 6) { - usb_audio_err(state->chip, - "unit %u: invalid UAC_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } csize = 4; channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; - if (hdr->bLength < 6 + csize) { - usb_audio_err(state->chip, - "unit %u: invalid UAC_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } } else { /* UAC_VERSION_3 */ struct uac3_feature_unit_descriptor *ftr = _ftr; - if (hdr->bLength < 7) { - usb_audio_err(state->chip, - "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } csize = 4; channels = (ftr->bLength - 7) / 4 - 1; bmaControls = ftr->bmaControls; - if (hdr->bLength < 7 + csize) { - usb_audio_err(state->chip, - "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n", - unitid); - return -EINVAL; - } } /* parse the source unit */ @@ -1988,6 +1970,31 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * Mixer Unit */ +/* check whether the given in/out overflows bmMixerControls matrix */ +static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc, + int protocol, int num_ins, int num_outs) +{ + u8 *hdr = (u8 *)desc; + u8 *c = uac_mixer_unit_bmControls(desc, protocol); + size_t rest; /* remaining bytes after bmMixerControls */ + + switch (protocol) { + case UAC_VERSION_1: + default: + rest = 1; /* iMixer */ + break; + case UAC_VERSION_2: + rest = 2; /* bmControls + iMixer */ + break; + case UAC_VERSION_3: + rest = 6; /* bmControls + wMixerDescrStr */ + break; + } + + /* overflow? */ + return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0]; +} + /* * build a mixer unit control * @@ -2030,7 +2037,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); - kfree(cval); + usb_mixer_elem_info_free(cval); return; } kctl->private_free = snd_usb_mixer_elem_free; @@ -2056,15 +2063,11 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_input_terminal_descriptor *d_v2 = raw_desc; - if (d_v2->bLength < sizeof(*d_v2)) - return -EINVAL; control = UAC2_TE_CONNECTOR; term_id = d_v2->bTerminalID; bmctls = le16_to_cpu(d_v2->bmControls); } else if (state->mixer->protocol == UAC_VERSION_3) { struct uac3_input_terminal_descriptor *d_v3 = raw_desc; - if (d_v3->bLength < sizeof(*d_v3)) - return -EINVAL; control = UAC3_TE_INSERTION; term_id = d_v3->bTerminalID; bmctls = le32_to_cpu(d_v3->bmControls); @@ -2116,6 +2119,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) return err; num_ins += iterm.channels; + if (mixer_bitmap_overflow(desc, state->mixer->protocol, + num_ins, num_outs)) + break; for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; @@ -2323,18 +2329,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, const char *name = extension_unit ? "Extension Unit" : "Processing Unit"; - if (desc->bLength < 13) { - usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); - return -EINVAL; - } - num_ins = desc->bNrInPins; - if (desc->bLength < 13 + num_ins || - desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { - usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); - return -EINVAL; - } - for (i = 0; i < num_ins; i++) { err = parse_audio_unit(state, desc->baSourceID[i]); if (err < 0) @@ -2425,7 +2420,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { - kfree(cval); + usb_mixer_elem_info_free(cval); return -ENOMEM; } kctl->private_free = snd_usb_mixer_elem_free; @@ -2563,7 +2558,7 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) if (kctl->private_data) { struct usb_mixer_elem_info *cval = kctl->private_data; num_ins = cval->max; - kfree(cval); + usb_mixer_elem_info_free(cval); kctl->private_data = NULL; } if (kctl->private_value) { @@ -2589,13 +2584,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, const struct usbmix_name_map *map; char **namelist; - if (desc->bLength < 5 || !desc->bNrInPins || - desc->bLength < 5 + desc->bNrInPins) { - usb_audio_err(state->chip, - "invalid SELECTOR UNIT descriptor %d\n", unitid); - return -EINVAL; - } - for (i = 0; i < desc->bNrInPins; i++) { err = parse_audio_unit(state, desc->baSourceID[i]); if (err < 0) @@ -2635,10 +2623,10 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, break; } - namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL); + namelist = kcalloc(desc->bNrInPins, sizeof(char *), GFP_KERNEL); if (!namelist) { - kfree(cval); - return -ENOMEM; + err = -ENOMEM; + goto error_cval; } #define MAX_ITEM_NAME_LEN 64 for (i = 0; i < desc->bNrInPins; i++) { @@ -2646,11 +2634,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, len = 0; namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL); if (!namelist[i]) { - while (i--) - kfree(namelist[i]); - kfree(namelist); - kfree(cval); - return -ENOMEM; + err = -ENOMEM; + goto error_name; } len = check_mapped_selector_name(state, unitid, i, namelist[i], MAX_ITEM_NAME_LEN); @@ -2664,11 +2649,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval); if (! kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); - for (i = 0; i < desc->bNrInPins; i++) - kfree(namelist[i]); - kfree(namelist); - kfree(cval); - return -ENOMEM; + err = -ENOMEM; + goto error_name; } kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; @@ -2714,6 +2696,14 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, usb_audio_dbg(state->chip, "[%d] SU [%s] items = %d\n", cval->head.id, kctl->id.name, desc->bNrInPins); return snd_usb_mixer_add_control(&cval->head, kctl); + + error_name: + for (i = 0; i < desc->bNrInPins; i++) + kfree(namelist[i]); + kfree(namelist); + error_cval: + usb_mixer_elem_info_free(cval); + return err; } /* @@ -2734,62 +2724,49 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return -EINVAL; } - if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { - switch (p1[2]) { - case UAC_INPUT_TERMINAL: - return parse_audio_input_terminal(state, unitid, p1); - case UAC_MIXER_UNIT: - return parse_audio_mixer_unit(state, unitid, p1); - case UAC2_CLOCK_SOURCE: - return parse_clock_source_unit(state, unitid, p1); - case UAC_SELECTOR_UNIT: - case UAC2_CLOCK_SELECTOR: - return parse_audio_selector_unit(state, unitid, p1); - case UAC_FEATURE_UNIT: - return parse_audio_feature_unit(state, unitid, p1); - case UAC1_PROCESSING_UNIT: - /* UAC2_EFFECT_UNIT has the same value */ - if (protocol == UAC_VERSION_1) - return parse_audio_processing_unit(state, unitid, p1); - else - return 0; /* FIXME - effect units not implemented yet */ - case UAC1_EXTENSION_UNIT: - /* UAC2_PROCESSING_UNIT_V2 has the same value */ - if (protocol == UAC_VERSION_1) - return parse_audio_extension_unit(state, unitid, p1); - else /* UAC_VERSION_2 */ - return parse_audio_processing_unit(state, unitid, p1); - case UAC2_EXTENSION_UNIT_V2: - return parse_audio_extension_unit(state, unitid, p1); - default: - usb_audio_err(state->chip, - "unit %u: unexpected type 0x%02x\n", unitid, p1[2]); - return -EINVAL; - } - } else { /* UAC_VERSION_3 */ - switch (p1[2]) { - case UAC_INPUT_TERMINAL: - return parse_audio_input_terminal(state, unitid, p1); - case UAC3_MIXER_UNIT: - return parse_audio_mixer_unit(state, unitid, p1); - case UAC3_CLOCK_SOURCE: - return parse_clock_source_unit(state, unitid, p1); - case UAC3_SELECTOR_UNIT: - case UAC3_CLOCK_SELECTOR: - return parse_audio_selector_unit(state, unitid, p1); - case UAC3_FEATURE_UNIT: - return parse_audio_feature_unit(state, unitid, p1); - case UAC3_EFFECT_UNIT: - return 0; /* FIXME - effect units not implemented yet */ - case UAC3_PROCESSING_UNIT: - return parse_audio_processing_unit(state, unitid, p1); - case UAC3_EXTENSION_UNIT: - return parse_audio_extension_unit(state, unitid, p1); - default: - usb_audio_err(state->chip, - "unit %u: unexpected type 0x%02x\n", unitid, p1[2]); - return -EINVAL; - } + if (!snd_usb_validate_audio_desc(p1, protocol)) { + usb_audio_dbg(state->chip, "invalid unit %d\n", unitid); + return 0; /* skip invalid unit */ + } + + switch (PTYPE(protocol, p1[2])) { + case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL): + case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL): + case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL): + return parse_audio_input_terminal(state, unitid, p1); + case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT): + return parse_audio_mixer_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE): + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE): + return parse_clock_source_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR): + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR): + return parse_audio_selector_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT): + return parse_audio_feature_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT): + return parse_audio_processing_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT): + return parse_audio_extension_unit(state, unitid, p1); + case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT): + return 0; /* FIXME - effect units not implemented yet */ + default: + usb_audio_err(state->chip, + "unit %u: unexpected type 0x%02x\n", + unitid, p1[2]); + return -EINVAL; } } @@ -3104,11 +3081,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { + if (!snd_usb_validate_audio_desc(p, mixer->protocol)) + continue; /* skip invalid descriptor */ + if (mixer->protocol == UAC_VERSION_1) { struct uac1_output_terminal_descriptor *desc = p; - if (desc->bLength < sizeof(*desc)) - continue; /* invalid descriptor? */ /* mark terminal ID as visited */ set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; @@ -3120,8 +3098,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } else if (mixer->protocol == UAC_VERSION_2) { struct uac2_output_terminal_descriptor *desc = p; - if (desc->bLength < sizeof(*desc)) - continue; /* invalid descriptor? */ /* mark terminal ID as visited */ set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; @@ -3147,8 +3123,6 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } else { /* UAC_VERSION_3 */ struct uac3_output_terminal_descriptor *desc = p; - if (desc->bLength < sizeof(*desc)) - continue; /* invalid descriptor? */ /* mark terminal ID as visited */ set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; @@ -3509,6 +3483,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } @@ -3536,6 +3512,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 394cd9107507..37e1b234c802 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,10 @@ struct usb_mixer_interface { struct media_mixer_ctl *media_mixer_ctl; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *mixer); + void (*private_suspend)(struct usb_mixer_interface *mixer); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..39e27ae6c597 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -32,6 +32,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -1155,17 +1156,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + int unitid = 12; /* SampleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); - if (cval) { + if (mixer->id_elems[unitid]) { + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); + break; } - break; } } @@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..7d460b1f1735 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2075 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +/* device_setup value to enable */ +#define SCARLETT2_ENABLE 0x01 + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT2_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT2_MIXER_MIN_DB -80 +#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2) +#define SCARLETT2_MIXER_MAX_DB 6 +#define SCARLETT2_MIXER_MAX_VALUE \ + ((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + struct scarlett2_usb_packet *req = NULL, *resp = NULL; + int err = 0; + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto error; + } + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto error; + } + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + mutex_unlock(&private->usb_mutex); +error: + kfree(req); + kfree(resp); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct usb_mixer_interface *mixer) +{ + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Delayed work to save config */ +static void scarlett2_config_save_work(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + + scarlett2_config_save(private->mixer); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Cancel any pending NVRAM save */ + cancel_delayed_work_sync(&private->work); + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT2_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_gain, -SCARLETT2_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT2_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_mixer, + SCARLETT2_MIXER_MIN_DB * 100, + SCARLETT2_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT2_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett2_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Cleanup/Suspend Callbacks ***/ + +static void scarlett2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} + +static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + if (cancel_delayed_work_sync(&private->work)) + scarlett2_config_save(private->mixer); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + struct usb_device *dev = mixer->chip->dev; + unsigned int pipe = usb_rcvintpipe(dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT); + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + if (snd_usb_pipe_sanity_check(dev, pipe)) + return -EINVAL; + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb(mixer->urb, dev, pipe, + transfer_buffer, SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!(mixer->chip->setup & SCARLETT2_ENABLE)) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; " + "use options snd_usb_audio device_setup=1 " + "to enable and report any issues to g@b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 75b96929f76c..ff5ab24f3bd1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; ifnum = 1; @@ -347,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ + /* BOSS Katana amplifiers do not need quirks */ + return 0; } if (attr == USB_ENDPOINT_SYNC_ASYNC && @@ -456,6 +460,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, } ep = get_endpoint(alts, 1)->bEndpointAddress; if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 0)->bSynchAddress != 0 && ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { dev_err(&dev->dev, diff --git a/sound/usb/power.c b/sound/usb/power.c index bd303a1ba1b7..606a2cb23eab 100644 --- a/sound/usb/power.c +++ b/sound/usb/power.c @@ -31,6 +31,8 @@ snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, struct uac3_power_domain_descriptor *pd_desc = p; int i; + if (!snd_usb_validate_audio_desc(p, UAC_VERSION_3)) + continue; for (i = 0; i < pd_desc->bNrEntities; i++) { if (pd_desc->baEntityID[i] == id) { pd->pd_id = pd_desc->bPowerDomainID; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e918ce346027..70c338f3ae24 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3534,5 +3534,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * PIONEER DJ DDJ-SX3 + * PCM is 12 channels out, 10 channels in @ 44.1 fixed + * interface 0, vendor class alt setting 1 for endpoints 5 and 0x86 + * The feedback for the output is the input. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0023), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..fbfde996fee7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -248,6 +248,9 @@ static int create_yamaha_midi_quirk(struct snd_usb_audio *chip, NULL, USB_MS_MIDI_OUT_JACK); if (!injd && !outjd) return -ENODEV; + if (!(injd && snd_usb_validate_midi_desc(injd)) || + !(outjd && snd_usb_validate_midi_desc(outjd))) + return -ENODEV; if (injd && (injd->bLength < 5 || (injd->bJackType != USB_MS_EMBEDDED && injd->bJackType != USB_MS_EXTERNAL))) @@ -1563,7 +1566,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct usb_interface *iface; /* Playback Designs */ - if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) { + if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && + USB_ID_PRODUCT(chip->usb_id) < 0x0110) { switch (fp->altsetting) { case 1: fp->dsd_dop = true; @@ -1580,9 +1584,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ - case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ - case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */ - case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) @@ -1596,7 +1597,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ - case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ @@ -1651,10 +1651,15 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, * from XMOS/Thesycon */ switch (USB_ID_VENDOR(chip->usb_id)) { - case 0x20b1: /* XMOS based devices */ case 0x152a: /* Thesycon devices */ + case 0x20b1: /* XMOS based devices */ + case 0x22d9: /* Oppo */ + case 0x23ba: /* Playback Designs */ case 0x25ce: /* Mytek devices */ + case 0x278b: /* Rotel? */ case 0x2ab6: /* T+A devices */ + case 0x3842: /* EVGA */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7ee9d17d0143..11785f9652ad 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -28,6 +28,14 @@ #include "power.h" #include "media.h" +static void audioformat_free(struct audioformat *fp) +{ + list_del(&fp->list); /* unlink for avoiding double-free */ + kfree(fp->rate_table); + kfree(fp->chmap); + kfree(fp); +} + /* * free a substream */ @@ -37,11 +45,8 @@ static void free_substream(struct snd_usb_substream *subs) if (!subs->num_formats) return; /* not initialized */ - list_for_each_entry_safe(fp, n, &subs->fmt_list, list) { - kfree(fp->rate_table); - kfree(fp->chmap); - kfree(fp); - } + list_for_each_entry_safe(fp, n, &subs->fmt_list, list) + audioformat_free(fp); kfree(subs->rate_list.list); kfree(subs->str_pd); snd_media_stream_delete(subs); @@ -627,16 +632,14 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, */ static void * snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id, bool uac23) + int terminal_id, int protocol) { struct uac2_input_terminal_descriptor *term = NULL; - size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) : - sizeof(struct uac_input_terminal_descriptor); while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_INPUT_TERMINAL))) { - if (term->bLength < minlen) + if (!snd_usb_validate_audio_desc(term, protocol)) continue; if (term->bTerminalID == terminal_id) return term; @@ -647,7 +650,7 @@ snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, static void * snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) + int terminal_id, int protocol) { /* OK to use with both UAC2 and UAC3 */ struct uac2_output_terminal_descriptor *term = NULL; @@ -655,8 +658,9 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_OUTPUT_TERMINAL))) { - if (term->bLength >= sizeof(*term) && - term->bTerminalID == terminal_id) + if (!snd_usb_validate_audio_desc(term, protocol)) + continue; + if (term->bTerminalID == terminal_id) return term; } @@ -731,7 +735,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, as->bTerminalLink, - false); + protocol); if (iterm) { num_channels = iterm->bNrChannels; chconfig = le16_to_cpu(iterm->wChannelConfig); @@ -767,7 +771,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, as->bTerminalLink, - true); + protocol); if (input_term) { clock = input_term->bCSourceID; if (!chconfig && (num_channels == input_term->bNrChannels)) @@ -776,7 +780,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, } output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + protocol); if (output_term) { clock = output_term->bCSourceID; goto found_clock; @@ -832,8 +837,7 @@ found_clock: /* ok, let's parse further... */ if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { - kfree(fp->rate_table); - kfree(fp); + audioformat_free(fp); return NULL; } @@ -1002,14 +1006,15 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, as->bTerminalLink, - true); + UAC_VERSION_3); if (input_term) { clock = input_term->bCSourceID; goto found_clock; } output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + UAC_VERSION_3); if (output_term) { clock = output_term->bCSourceID; goto found_clock; @@ -1043,8 +1048,7 @@ found_clock: pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { - kfree(fp->rate_table); - kfree(fp); + audioformat_free(fp); return NULL; } pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ? @@ -1063,9 +1067,7 @@ found_clock: /* ok, let's parse further... */ if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { kfree(pd); - kfree(fp->chmap); - kfree(fp->rate_table); - kfree(fp); + audioformat_free(fp); return NULL; } } @@ -1076,7 +1078,9 @@ found_clock: return fp; } -int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, + int iface_no, + bool *has_non_pcm, bool non_pcm) { struct usb_device *dev; struct usb_interface *iface; @@ -1177,6 +1181,16 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) else if (IS_ERR(fp)) return PTR_ERR(fp); + if (fp->fmt_type != UAC_FORMAT_TYPE_I) + *has_non_pcm = true; + if ((fp->fmt_type == UAC_FORMAT_TYPE_I) == non_pcm) { + audioformat_free(fp); + kfree(pd); + fp = NULL; + pd = NULL; + continue; + } + dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); if (protocol == UAC_VERSION_3) err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd); @@ -1184,11 +1198,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { - list_del(&fp->list); /* unlink for avoiding double-free */ + audioformat_free(fp); kfree(pd); - kfree(fp->rate_table); - kfree(fp->chmap); - kfree(fp); return err; } /* try to set the interface... */ @@ -1199,3 +1210,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return 0; } +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + int err; + bool has_non_pcm = false; + + /* parse PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false); + if (err < 0) + return err; + + if (has_non_pcm) { + /* parse non-PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true); + if (err < 0) + return err; + } + + return 0; +} + diff --git a/sound/usb/validate.c b/sound/usb/validate.c new file mode 100644 index 000000000000..3c8f73a0eb12 --- /dev/null +++ b/sound/usb/validate.c @@ -0,0 +1,332 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// Validation of USB-audio class descriptors +// + +#include <linux/init.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> +#include <linux/usb/audio-v3.h> +#include <linux/usb/midi.h> +#include "usbaudio.h" +#include "helper.h" + +struct usb_desc_validator { + unsigned char protocol; + unsigned char type; + bool (*func)(const void *p, const struct usb_desc_validator *v); + size_t size; +}; + +#define UAC_VERSION_ALL (unsigned char)(-1) + +/* UAC1 only */ +static bool validate_uac1_header(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac1_ac_header_descriptor *d = p; + + return d->bLength >= sizeof(*d) && + d->bLength >= sizeof(*d) + d->bInCollection; +} + +/* for mixer unit; covering all UACs */ +static bool validate_mixer_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac_mixer_unit_descriptor *d = p; + size_t len; + + if (d->bLength < sizeof(*d) || !d->bNrInPins) + return false; + len = sizeof(*d) + d->bNrInPins; + /* We can't determine the bitmap size only from this unit descriptor, + * so just check with the remaining length. + * The actual bitmap is checked at mixer unit parser. + */ + switch (v->protocol) { + case UAC_VERSION_1: + default: + len += 2 + 1; /* wChannelConfig, iChannelNames */ + /* bmControls[n*m] */ + len += 1; /* iMixer */ + break; + case UAC_VERSION_2: + len += 4 + 1; /* bmChannelConfig, iChannelNames */ + /* bmMixerControls[n*m] */ + len += 1 + 1; /* bmControls, iMixer */ + break; + case UAC_VERSION_3: + len += 2; /* wClusterDescrID */ + /* bmMixerControls[n*m] */ + break; + } + return d->bLength >= len; +} + +/* both for processing and extension units; covering all UACs */ +static bool validate_processing_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac_processing_unit_descriptor *d = p; + const unsigned char *hdr = p; + size_t len, m; + + if (d->bLength < sizeof(*d)) + return false; + len = d->bLength < sizeof(*d) + d->bNrInPins; + if (d->bLength < len) + return false; + switch (v->protocol) { + case UAC_VERSION_1: + default: + /* bNrChannels, wChannelConfig, iChannelNames, bControlSize */ + len += 1 + 2 + 1 + 1; + if (d->bLength < len) /* bControlSize */ + return false; + m = hdr[len]; + len += 1 + m + 1; /* bControlSize, bmControls, iProcessing */ + break; + case UAC_VERSION_2: + /* bNrChannels, bmChannelConfig, iChannelNames */ + len += 1 + 4 + 1; + if (v->type == UAC2_PROCESSING_UNIT_V2) + len += 2; /* bmControls -- 2 bytes for PU */ + else + len += 1; /* bmControls -- 1 byte for EU */ + len += 1; /* iProcessing */ + break; + case UAC_VERSION_3: + /* wProcessingDescrStr, bmControls */ + len += 2 + 4; + break; + } + if (d->bLength < len) + return false; + + switch (v->protocol) { + case UAC_VERSION_1: + default: + if (v->type == UAC1_EXTENSION_UNIT) + return true; /* OK */ + switch (d->wProcessType) { + case UAC_PROCESS_UP_DOWNMIX: + case UAC_PROCESS_DOLBY_PROLOGIC: + if (d->bLength < len + 1) /* bNrModes */ + return false; + m = hdr[len]; + len += 1 + m * 2; /* bNrModes, waModes(n) */ + break; + default: + break; + } + break; + case UAC_VERSION_2: + if (v->type == UAC2_EXTENSION_UNIT_V2) + return true; /* OK */ + switch (d->wProcessType) { + case UAC2_PROCESS_UP_DOWNMIX: + case UAC2_PROCESS_DOLBY_PROLOCIC: /* SiC! */ + if (d->bLength < len + 1) /* bNrModes */ + return false; + m = hdr[len]; + len += 1 + m * 4; /* bNrModes, daModes(n) */ + break; + default: + break; + } + break; + case UAC_VERSION_3: + if (v->type == UAC3_EXTENSION_UNIT) { + len += 2; /* wClusterDescrID */ + break; + } + switch (d->wProcessType) { + case UAC3_PROCESS_UP_DOWNMIX: + if (d->bLength < len + 1) /* bNrModes */ + return false; + m = hdr[len]; + len += 1 + m * 2; /* bNrModes, waClusterDescrID(n) */ + break; + case UAC3_PROCESS_MULTI_FUNCTION: + len += 2 + 4; /* wClusterDescrID, bmAlgorighms */ + break; + default: + break; + } + break; + } + if (d->bLength < len) + return false; + + return true; +} + +/* both for selector and clock selector units; covering all UACs */ +static bool validate_selector_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac_selector_unit_descriptor *d = p; + size_t len; + + if (d->bLength < sizeof(*d)) + return false; + len = sizeof(*d) + d->bNrInPins; + switch (v->protocol) { + case UAC_VERSION_1: + default: + len += 1; /* iSelector */ + break; + case UAC_VERSION_2: + len += 1 + 1; /* bmControls, iSelector */ + break; + case UAC_VERSION_3: + len += 4 + 2; /* bmControls, wSelectorDescrStr */ + break; + } + return d->bLength >= len; +} + +static bool validate_uac1_feature_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac_feature_unit_descriptor *d = p; + + if (d->bLength < sizeof(*d) || !d->bControlSize) + return false; + /* at least bmaControls(0) for master channel + iFeature */ + return d->bLength >= sizeof(*d) + d->bControlSize + 1; +} + +static bool validate_uac2_feature_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac2_feature_unit_descriptor *d = p; + + if (d->bLength < sizeof(*d)) + return false; + /* at least bmaControls(0) for master channel + iFeature */ + return d->bLength >= sizeof(*d) + 4 + 1; +} + +static bool validate_uac3_feature_unit(const void *p, + const struct usb_desc_validator *v) +{ + const struct uac3_feature_unit_descriptor *d = p; + + if (d->bLength < sizeof(*d)) + return false; + /* at least bmaControls(0) for master channel + wFeatureDescrStr */ + return d->bLength >= sizeof(*d) + 4 + 2; +} + +static bool validate_midi_out_jack(const void *p, + const struct usb_desc_validator *v) +{ + const struct usb_midi_out_jack_descriptor *d = p; + + return d->bLength >= sizeof(*d) && + d->bLength >= sizeof(*d) + d->bNrInputPins * 2; +} + +#define FIXED(p, t, s) { .protocol = (p), .type = (t), .size = sizeof(s) } +#define FUNC(p, t, f) { .protocol = (p), .type = (t), .func = (f) } + +static struct usb_desc_validator audio_validators[] = { + /* UAC1 */ + FUNC(UAC_VERSION_1, UAC_HEADER, validate_uac1_header), + FIXED(UAC_VERSION_1, UAC_INPUT_TERMINAL, + struct uac_input_terminal_descriptor), + FIXED(UAC_VERSION_1, UAC_OUTPUT_TERMINAL, + struct uac1_output_terminal_descriptor), + FUNC(UAC_VERSION_1, UAC_MIXER_UNIT, validate_mixer_unit), + FUNC(UAC_VERSION_1, UAC_SELECTOR_UNIT, validate_selector_unit), + FUNC(UAC_VERSION_1, UAC_FEATURE_UNIT, validate_uac1_feature_unit), + FUNC(UAC_VERSION_1, UAC1_PROCESSING_UNIT, validate_processing_unit), + FUNC(UAC_VERSION_1, UAC1_EXTENSION_UNIT, validate_processing_unit), + + /* UAC2 */ + FIXED(UAC_VERSION_2, UAC_HEADER, struct uac2_ac_header_descriptor), + FIXED(UAC_VERSION_2, UAC_INPUT_TERMINAL, + struct uac2_input_terminal_descriptor), + FIXED(UAC_VERSION_2, UAC_OUTPUT_TERMINAL, + struct uac2_output_terminal_descriptor), + FUNC(UAC_VERSION_2, UAC_MIXER_UNIT, validate_mixer_unit), + FUNC(UAC_VERSION_2, UAC_SELECTOR_UNIT, validate_selector_unit), + FUNC(UAC_VERSION_2, UAC_FEATURE_UNIT, validate_uac2_feature_unit), + /* UAC_VERSION_2, UAC2_EFFECT_UNIT: not implemented yet */ + FUNC(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2, validate_processing_unit), + FUNC(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2, validate_processing_unit), + FIXED(UAC_VERSION_2, UAC2_CLOCK_SOURCE, + struct uac_clock_source_descriptor), + FUNC(UAC_VERSION_2, UAC2_CLOCK_SELECTOR, validate_selector_unit), + FIXED(UAC_VERSION_2, UAC2_CLOCK_MULTIPLIER, + struct uac_clock_multiplier_descriptor), + /* UAC_VERSION_2, UAC2_SAMPLE_RATE_CONVERTER: not implemented yet */ + + /* UAC3 */ + FIXED(UAC_VERSION_2, UAC_HEADER, struct uac3_ac_header_descriptor), + FIXED(UAC_VERSION_3, UAC_INPUT_TERMINAL, + struct uac3_input_terminal_descriptor), + FIXED(UAC_VERSION_3, UAC_OUTPUT_TERMINAL, + struct uac3_output_terminal_descriptor), + /* UAC_VERSION_3, UAC3_EXTENDED_TERMINAL: not implemented yet */ + FUNC(UAC_VERSION_3, UAC3_MIXER_UNIT, validate_mixer_unit), + FUNC(UAC_VERSION_3, UAC3_SELECTOR_UNIT, validate_selector_unit), + FUNC(UAC_VERSION_3, UAC_FEATURE_UNIT, validate_uac3_feature_unit), + /* UAC_VERSION_3, UAC3_EFFECT_UNIT: not implemented yet */ + FUNC(UAC_VERSION_3, UAC3_PROCESSING_UNIT, validate_processing_unit), + FUNC(UAC_VERSION_3, UAC3_EXTENSION_UNIT, validate_processing_unit), + FIXED(UAC_VERSION_3, UAC3_CLOCK_SOURCE, + struct uac3_clock_source_descriptor), + FUNC(UAC_VERSION_3, UAC3_CLOCK_SELECTOR, validate_selector_unit), + FIXED(UAC_VERSION_3, UAC3_CLOCK_MULTIPLIER, + struct uac3_clock_multiplier_descriptor), + /* UAC_VERSION_3, UAC3_SAMPLE_RATE_CONVERTER: not implemented yet */ + /* UAC_VERSION_3, UAC3_CONNECTORS: not implemented yet */ + { } /* terminator */ +}; + +static struct usb_desc_validator midi_validators[] = { + FIXED(UAC_VERSION_ALL, USB_MS_HEADER, + struct usb_ms_header_descriptor), + FIXED(UAC_VERSION_ALL, USB_MS_MIDI_IN_JACK, + struct usb_midi_in_jack_descriptor), + FUNC(UAC_VERSION_ALL, USB_MS_MIDI_OUT_JACK, + validate_midi_out_jack), + { } /* terminator */ +}; + + +/* Validate the given unit descriptor, return true if it's OK */ +static bool validate_desc(unsigned char *hdr, int protocol, + const struct usb_desc_validator *v) +{ + if (hdr[1] != USB_DT_CS_INTERFACE) + return true; /* don't care */ + + for (; v->type; v++) { + if (v->type == hdr[2] && + (v->protocol == UAC_VERSION_ALL || + v->protocol == protocol)) { + if (v->func) + return v->func(hdr, v); + /* check for the fixed size */ + return hdr[0] >= v->size; + } + } + + return true; /* not matching, skip validation */ +} + +bool snd_usb_validate_audio_desc(void *p, int protocol) +{ + return validate_desc(p, protocol, audio_validators); +} + +bool snd_usb_validate_midi_desc(void *p) +{ + return validate_desc(p, UAC_VERSION_1, midi_validators); +} + |