diff options
Diffstat (limited to 'sound')
160 files changed, 3534 insertions, 1675 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ec9e7866177f..446c00bd908b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1015,6 +1015,60 @@ int snd_interval_list(struct snd_interval *i, unsigned int count, EXPORT_SYMBOL(snd_interval_list); +/** + * snd_interval_ranges - refine the interval value from the list of ranges + * @i: the interval value to refine + * @count: the number of elements in the list of ranges + * @ranges: the ranges list + * @mask: the bit-mask to evaluate + * + * Refines the interval value from the list of ranges. + * When mask is non-zero, only the elements corresponding to bit 1 are + * evaluated. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_interval_ranges(struct snd_interval *i, unsigned int count, + const struct snd_interval *ranges, unsigned int mask) +{ + unsigned int k; + struct snd_interval range_union; + struct snd_interval range; + + if (!count) { + snd_interval_none(i); + return -EINVAL; + } + snd_interval_any(&range_union); + range_union.min = UINT_MAX; + range_union.max = 0; + for (k = 0; k < count; k++) { + if (mask && !(mask & (1 << k))) + continue; + snd_interval_copy(&range, &ranges[k]); + if (snd_interval_refine(&range, i) < 0) + continue; + if (snd_interval_empty(&range)) + continue; + + if (range.min < range_union.min) { + range_union.min = range.min; + range_union.openmin = 1; + } + if (range.min == range_union.min && !range.openmin) + range_union.openmin = 0; + if (range.max > range_union.max) { + range_union.max = range.max; + range_union.openmax = 1; + } + if (range.max == range_union.max && !range.openmax) + range_union.openmax = 0; + } + return snd_interval_refine(i, &range_union); +} +EXPORT_SYMBOL(snd_interval_ranges); + static int snd_interval_step(struct snd_interval *i, unsigned int step) { unsigned int n; @@ -1221,6 +1275,37 @@ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, EXPORT_SYMBOL(snd_pcm_hw_constraint_list); +static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hw_constraint_ranges *r = rule->private; + return snd_interval_ranges(hw_param_interval(params, rule->var), + r->count, r->ranges, r->mask); +} + + +/** + * snd_pcm_hw_constraint_ranges - apply list of range constraints to a parameter + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the list of range constraints + * @r: ranges + * + * Apply the list of range constraints to an interval parameter. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_ranges *r) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_ranges, (void *)r, + var, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); + static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index ec667f158f19..5d905d90d504 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -82,36 +82,6 @@ struct snd_seq_dummy_port { static int my_client = -1; /* - * unuse callback - send ALL_SOUNDS_OFF and RESET_CONTROLLERS events - * to subscribers. - * Note: this callback is called only after all subscribers are removed. - */ -static int -dummy_unuse(void *private_data, struct snd_seq_port_subscribe *info) -{ - struct snd_seq_dummy_port *p; - int i; - struct snd_seq_event ev; - - p = private_data; - memset(&ev, 0, sizeof(ev)); - if (p->duplex) - ev.source.port = p->connect; - else - ev.source.port = p->port; - ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; - ev.type = SNDRV_SEQ_EVENT_CONTROLLER; - for (i = 0; i < 16; i++) { - ev.data.control.channel = i; - ev.data.control.param = MIDI_CTL_ALL_SOUNDS_OFF; - snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0); - ev.data.control.param = MIDI_CTL_RESET_CONTROLLERS; - snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0); - } - return 0; -} - -/* * event input callback - just redirect events to subscribers */ static int @@ -175,7 +145,6 @@ create_port(int idx, int type) | SNDRV_SEQ_PORT_TYPE_PORT; memset(&pcb, 0, sizeof(pcb)); pcb.owner = THIS_MODULE; - pcb.unuse = dummy_unuse; pcb.event_input = dummy_input; pcb.private_free = dummy_free; pcb.private_data = rec; diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 3badc70124ab..0d580186ef1a 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -21,7 +21,19 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */ +/* + * Nominally 3125 bytes/second, but the MIDI port's clock might be + * 1% too slow, and the bus clock 100 ppm too fast. + */ +#define MIDI_BYTES_PER_SECOND 3093 + +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 @@ -78,8 +90,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->sync_slave = NULL; - s->rx_blocks_for_midi = UINT_MAX; - return 0; } EXPORT_SYMBOL(amdtp_stream_init); @@ -222,6 +232,14 @@ sfc_found: for (i = 0; i < pcm_channels; i++) s->pcm_positions[i] = i; s->midi_position = s->pcm_channels; + + /* + * We do not know the actual MIDI FIFO size of most devices. Just + * assume two bytes, i.e., one byte can be received over the bus while + * the previous one is transmitted over MIDI. + * (The value here is adjusted for midi_ratelimit_per_packet().) + */ + s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1; } EXPORT_SYMBOL(amdtp_stream_set_parameters); @@ -463,6 +481,36 @@ static void amdtp_fill_pcm_silence(struct amdtp_stream *s, } } +/* + * To avoid sending MIDI bytes at too high a rate, assume that the receiving + * device has a FIFO, and track how much it is filled. This values increases + * by one whenever we send one byte in a packet, but the FIFO empties at + * a constant rate independent of our packet rate. One packet has syt_interval + * samples, so the number of bytes that empty out of the FIFO, per packet(!), + * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing + * fractional values, the values in midi_fifo_used[] are measured in bytes + * multiplied by the sample rate. + */ +static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port) +{ + int used; + + used = s->midi_fifo_used[port]; + if (used == 0) /* common shortcut */ + return true; + + used -= MIDI_BYTES_PER_SECOND * s->syt_interval; + used = max(used, 0); + s->midi_fifo_used[port] = used; + + return used < s->midi_fifo_limit; +} + +static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port) +{ + s->midi_fifo_used[port] += amdtp_rate_table[s->sfc]; +} + static void amdtp_fill_midi(struct amdtp_stream *s, __be32 *buffer, unsigned int frames) { @@ -470,16 +518,21 @@ static void amdtp_fill_midi(struct amdtp_stream *s, u8 *b; for (f = 0; f < frames; f++) { - buffer[s->midi_position] = 0; b = (u8 *)&buffer[s->midi_position]; port = (s->data_block_counter + f) % 8; - if ((f >= s->rx_blocks_for_midi) || - (s->midi[port] == NULL) || - (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0)) - b[0] = 0x80; - else + if (f < MAX_MIDI_RX_BLOCKS && + midi_ratelimit_per_packet(s, port) && + s->midi[port] != NULL && + snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) { + midi_rate_use_one_byte(s, port); b[0] = 0x81; + } else { + b[0] = 0x80; + b[1] = 0; + } + b[2] = 0; + b[3] = 0; buffer += s->data_block_quadlets; } diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index e6e8926275b0..8a03a91e728b 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -148,13 +148,12 @@ struct amdtp_stream { bool double_pcm_frames; struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; + int midi_fifo_limit; + int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; /* quirk: fixed interval of dbc between previos/current packets. */ unsigned int tx_dbc_interval; - /* quirk: the first count of data blocks in an rx packet for MIDI */ - unsigned int rx_blocks_for_midi; - bool callbacked; wait_queue_head_t callback_wait; struct amdtp_stream *sync_slave; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 1aab0a32870c..0ebcabfdc7ce 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) amdtp_stream_destroy(&bebob->rx_stream); destroy_both_connections(bebob); } - /* - * The firmware for these devices ignore MIDI messages in more than - * first 8 data blocks of an received AMDTP packet. - */ - if (bebob->spec == &maudio_fw410_spec || - bebob->spec == &maudio_special_spec) - bebob->rx_stream.rx_blocks_for_midi = 8; end: return err; } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index b985fc5ebdc6..4f440e163667 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) destroy_stream(efw, &efw->tx_stream); goto end; } - /* - * Fireworks ignores MIDI messages in more than first 8 data - * blocks of an received AMDTP packet. - */ - efw->rx_stream.rx_blocks_for_midi = 8; /* set IEC61883 compliant mode (actually not fully compliant...) */ err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883); diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 255dabc6fc33..2a85e4209f0b 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -124,7 +124,7 @@ copy_resp_to_buf(struct snd_efw *efw, void *data, size_t length, int *rcode) spin_lock_irq(&efw->lock); t = (struct snd_efw_transaction *)data; - length = min_t(size_t, t->length * sizeof(t->length), length); + length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length); if (efw->push_ptr < efw->pull_ptr) capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8276a743e22e..0cfc9c8c4b4e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2bf0b568e3de..d426a0bd6a5f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -299,6 +299,9 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_SKYLAKE \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ @@ -2027,7 +2030,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fdf4338..166e3e84b963 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5f13d2d18079..b422e406a9cb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3353,6 +3353,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3413,6 +3414,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); +MODULE_ALIAS("snd-hda-codec-id:10de0072"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4f6413e01c13..605d14003d25 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,9 +568,9 @@ static void stac_store_hints(struct hda_codec *codec) spec->gpio_mask; } if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_data &= spec->gpio_mask; if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) spec->eapd_mask &= spec->gpio_mask; if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860f7230..4c23381727a1 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index fb3878312bf8..1579e994acf8 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 33fb3bb133df..b8e7bad05eb1 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -105,13 +105,11 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, return ret; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config->dst_addr = ssc->phybase + SSC_THR; - slave_config->dst_maxburst = 1; - } else { - slave_config->src_addr = ssc->phybase + SSC_RHR; - slave_config->src_maxburst = 1; - } + slave_config->dst_addr = ssc->phybase + SSC_THR; + slave_config->dst_maxburst = 1; + + slave_config->src_addr = ssc->phybase + SSC_RHR; + slave_config->src_maxburst = 1; prtd->dma_intr_handler = atmel_pcm_dma_irq; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e2a25d..fb0b7e8b08ff 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -204,6 +204,13 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC to keep it at a clean status */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; @@ -250,11 +257,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; if (dma_params != NULL) { - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); - pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", - (dir ? "receive" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - dma_params->ssc = NULL; dma_params->substream = NULL; ssc_p->dma_params[dir] = NULL; @@ -266,10 +268,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { - /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dai: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - free_irq(ssc_p->ssc->irq, ssc_p); ssc_p->initialized = 0; } @@ -280,6 +278,10 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; } spin_unlock_irq(&ssc_p->lock); + + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dai: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); } @@ -348,7 +350,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -451,25 +452,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is - * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - + /* I2S format, CODEC supplies BCLK and LRC clocks. */ rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +464,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +481,55 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFS: + /* I2S format, CODEC supplies BCLK, SSC supplies LRCLK. */ + if (bits > 16 && !ssc->pdata->has_fslen_ext) { + dev_err(dai->dev, + "sample size %d is too large for SSC device\n", + bits); + return -EINVAL; + } + + fslen_ext = (bits - 1) / 16; + fslen = (bits - 1) % 16; + + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_PIN : SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, fslen) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_CLOCK : SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_NEGATIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, fslen) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); @@ -512,7 +546,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -527,7 +561,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | SSC_BF(TCMR_STTDLY, 1) | SSC_BF(TCMR_START, SSC_START_RISING_RF) - | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | SSC_BF(TCMR_CKS, SSC_CKS_DIV); @@ -545,10 +579,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, /* * DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks. * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is - * generated from the transmit clock. - * * Data is transferred on first BCLK after LRC pulse rising * edge.If stereo, the right channel data is contiguous with * the left channel data. @@ -556,7 +586,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); @@ -597,23 +627,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr, rfmr, tcmr, tfmr); if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - pr_debug("atmel_ssc_dai: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + if (!ssc_p->ssc->pdata->use_dma) { + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + } ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, ssc_p->name, ssc_p); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 66b66d0e7514..f5ad214663f9 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -47,7 +47,6 @@ #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/hardware.h> #include "../codecs/wm8731.h" #include "atmel-pcm.h" @@ -64,33 +63,6 @@ static struct clk *mclk; -static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops at91sam9g20ek_ops = { - .hw_params = at91sam9g20ek_hw_params, -}; - static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -173,7 +145,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .init = at91sam9g20ek_wm8731_init, .platform_name = "at91rm9200_ssc.0", .codec_name = "wm8731.0-001b", - .ops = &at91sam9g20ek_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card snd_soc_at91sam9g20ek = { diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a747ac0b399f..c75995f2779c 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -91,27 +91,12 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; /* WM8731 has its own 12MHz crystal */ snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, 12000000, SND_SOC_CLOCK_IN); - /* codec is bitclock and lrclk master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - goto out; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - goto out; - - ret = 0; -out: - return ret; + return 0; } static struct snd_soc_ops db1200_i2s_wm8731_ops = { @@ -125,6 +110,8 @@ static struct snd_soc_dai_link db1200_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.1", .platform_name = "au1xpsc-pcm.1", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index b06b8d8128c6..dd94fea72d5d 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -315,11 +315,6 @@ static struct snd_pcm_ops au1xpsc_pcm_ops = { .pointer = au1xpsc_pcm_pointer, }; -static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; @@ -335,7 +330,6 @@ static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_platform_driver au1xpsc_soc_platform = { .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, - .pcm_free = au1xpsc_pcm_free_dma_buffers, }; static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 6ffaaff469c7..24cc7f40d87a 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -287,11 +287,6 @@ static struct snd_pcm_ops alchemy_pcm_ops = { .pointer = alchemy_pcm_pointer, }; -static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; @@ -305,7 +300,6 @@ static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_platform_driver alchemy_pcm_soc_platform = { .ops = &alchemy_pcm_ops, .pcm_new = alchemy_pcm_new, - .pcm_free = alchemy_pcm_free_dma_buffers, }; static int alchemy_pcm_drvprobe(struct platform_device *pdev) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index a2bf27f4baab..a0f265327fdf 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -386,7 +386,7 @@ static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol, static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); /* * In order to avoid current on the load, mute power-on and power-off @@ -403,7 +403,7 @@ static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, static int pm860x_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int dac = 0; int data; diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 387530b0b0fd..17c953595660 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -333,8 +333,8 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); - /* powerdown dac, dac in tdm mode */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x41); + /* dac in tdm mode */ + regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); /* high-pass filter enable */ regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 686cacb0e835..632e89f793a7 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -163,7 +163,7 @@ static const struct snd_kcontrol_new ak4671_snd_controls[] = { static int ak4671_out2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index bdf8c5ac8ca4..0e357996864b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -55,18 +55,20 @@ static inline int alc5623_reset(struct snd_soc_codec *codec) static int amp_mixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + /* to power-on/off class-d amp generators/speaker */ /* need to write to 'index-46h' register : */ /* so write index num (here 0x46) to reg 0x6a */ /* and then 0xffff/0 to reg 0x6c */ - snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); + snd_soc_write(codec, ALC5623_HID_CTRL_INDEX, 0x46); switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); + snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0xFFFF); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); + snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0); break; } diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d1fdbc266631..db3283abbe18 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -116,18 +116,20 @@ static inline int alc5632_reset(struct regmap *map) static int amp_mixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + /* to power-on/off class-d amp generators/speaker */ /* need to write to 'index-46h' register : */ /* so write index num (here 0x46) to reg 0x6a */ /* and then 0xffff/0 to reg 0x6c */ - snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + snd_soc_write(codec, ALC5632_HID_CTRL_INDEX, 0x46); switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + snd_soc_write(codec, ALC5632_HID_CTRL_DATA, 0xFFFF); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + snd_soc_write(codec, ALC5632_HID_CTRL_DATA, 0); break; } @@ -1066,7 +1068,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_device_alc5632 = { +static const struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, @@ -1080,7 +1082,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), }; -static struct regmap_config alc5632_regmap = { +static const struct regmap_config alc5632_regmap = { .reg_bits = 8, .val_bits = 16, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9550d7433ad0..29202610dd0d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -84,7 +84,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); bool manual_ena = false; @@ -692,7 +692,8 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); unsigned int reg; if (w->shift % 2) @@ -705,25 +706,25 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, priv->in_pending++; break; case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE, 0); /* If this is the last input pending then allow VU */ priv->in_pending--; if (priv->in_pending == 0) { msleep(1); - arizona_in_set_vu(w->codec, 1); + arizona_in_set_vu(codec, 1); } break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, reg, + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU); break; case SND_SOC_DAPM_POST_PMD: /* Disable volume updates if no inputs are enabled */ - reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES); + reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) - arizona_in_set_vu(w->codec, 0); + arizona_in_set_vu(codec, 0); } return 0; @@ -734,7 +735,25 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case ARIZONA_OUT1L_ENA_SHIFT: + case ARIZONA_OUT1R_ENA_SHIFT: + case ARIZONA_OUT2L_ENA_SHIFT: + case ARIZONA_OUT2R_ENA_SHIFT: + case ARIZONA_OUT3L_ENA_SHIFT: + case ARIZONA_OUT3R_ENA_SHIFT: + priv->out_up_pending++; + priv->out_up_delay += 17; + break; + default: + break; + } + break; case SND_SOC_DAPM_POST_PMU: switch (w->shift) { case ARIZONA_OUT1L_ENA_SHIFT: @@ -743,13 +762,50 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT2R_ENA_SHIFT: case ARIZONA_OUT3L_ENA_SHIFT: case ARIZONA_OUT3R_ENA_SHIFT: - msleep(17); + priv->out_up_pending--; + if (!priv->out_up_pending) { + msleep(priv->out_up_delay); + priv->out_up_delay = 0; + } break; default: break; } break; + case SND_SOC_DAPM_PRE_PMD: + switch (w->shift) { + case ARIZONA_OUT1L_ENA_SHIFT: + case ARIZONA_OUT1R_ENA_SHIFT: + case ARIZONA_OUT2L_ENA_SHIFT: + case ARIZONA_OUT2R_ENA_SHIFT: + case ARIZONA_OUT3L_ENA_SHIFT: + case ARIZONA_OUT3R_ENA_SHIFT: + priv->out_down_pending++; + priv->out_down_delay++; + break; + default: + break; + } + break; + case SND_SOC_DAPM_POST_PMD: + switch (w->shift) { + case ARIZONA_OUT1L_ENA_SHIFT: + case ARIZONA_OUT1R_ENA_SHIFT: + case ARIZONA_OUT2L_ENA_SHIFT: + case ARIZONA_OUT2R_ENA_SHIFT: + case ARIZONA_OUT3L_ENA_SHIFT: + case ARIZONA_OUT3R_ENA_SHIFT: + priv->out_down_pending--; + if (!priv->out_down_pending) { + msleep(priv->out_down_delay); + priv->out_down_delay = 0; + } + break; + default: + break; + } + break; } return 0; @@ -760,7 +816,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; @@ -772,6 +829,9 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: val = 0; break; + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMD: + return arizona_out_ev(w, kcontrol, event); default: return -EINVAL; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 942cfb197b6d..11ff899b0272 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -77,6 +77,11 @@ struct arizona_priv { int num_inputs; unsigned int in_pending; + unsigned int out_up_pending; + unsigned int out_up_delay; + unsigned int out_down_pending; + unsigned int out_down_delay; + unsigned int spk_ena:2; unsigned int spk_ena_pending:1; }; diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 5075bf0a7276..e7238b8904bc 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -86,5 +86,5 @@ static struct platform_driver bt_sco_driver = { module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); -MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); +MODULE_DESCRIPTION("ASoC generic bluetooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index ec55c590afd0..f2b8aad21274 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -264,7 +264,7 @@ static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val); } -static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { .set_sysclk = cs35l32_codec_set_sysclk, .dapm_widgets = cs35l32_dapm_widgets, @@ -288,7 +288,7 @@ static const struct reg_default cs35l32_monitor_patch[] = { { 0x00, 0x00 }, }; -static struct regmap_config cs35l32_regmap = { +static const struct regmap_config cs35l32_regmap = { .reg_bits = 8, .val_bits = 8, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 35fbef743fbe..1589e7a881d8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1103,7 +1103,7 @@ static int cs42l52_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { .probe = cs42l52_probe, .remove = cs42l52_remove, .set_bias_level = cs42l52_set_bias_level, @@ -1130,7 +1130,7 @@ static const struct reg_default cs42l52_threshold_patch[] = { }; -static struct regmap_config cs42l52_regmap = { +static const struct regmap_config cs42l52_regmap = { .reg_bits = 8, .val_bits = 8, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 2ddc7ac10ad7..cbc654fe48c7 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1164,7 +1164,7 @@ static int cs42l56_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { .probe = cs42l56_probe, .remove = cs42l56_remove, .set_bias_level = cs42l56_set_bias_level, @@ -1179,7 +1179,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { .num_controls = ARRAY_SIZE(cs42l56_snd_controls), }; -static struct regmap_config cs42l56_regmap = { +static const struct regmap_config cs42l56_regmap = { .reg_bits = 8, .val_bits = 8, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7c55537c69cf..8ecedba79606 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1347,7 +1347,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .probe = cs42l73_probe, .set_bias_level = cs42l73_set_bias_level, .suspend_bias_off = true, @@ -1361,7 +1361,7 @@ static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .num_controls = ARRAY_SIZE(cs42l73_snd_controls), }; -static struct regmap_config cs42l73_regmap = { +static const struct regmap_config cs42l73_regmap = { .reg_bits = 8, .val_bits = 8, diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 61b2f9a2eef1..ffe96175a8a5 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -609,7 +609,7 @@ static const struct snd_kcontrol_new da732x_snd_controls[] = { static int da732x_adc_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -663,7 +663,7 @@ static int da732x_adc_event(struct snd_soc_dapm_widget *w, static int da732x_out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 7e73fa4b3183..8fb445f33f6f 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -32,7 +32,7 @@ static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct pcm3008_setup_data *setup = codec->dev->platform_data; gpio_set_value_cansleep(setup->pdda_pin, @@ -45,7 +45,7 @@ static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct pcm3008_setup_data *setup = codec->dev->platform_data; gpio_set_value_cansleep(setup->pdad_pin, diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index d0547fa275fc..dcdfac0ffeb1 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -46,6 +46,8 @@ static int pcm512x_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id pcm512x_i2c_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { } }; MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); @@ -53,6 +55,8 @@ MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index f297058c0038..7b64a9cef704 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -43,6 +43,8 @@ static int pcm512x_spi_remove(struct spi_device *spi) static const struct spi_device_id pcm512x_spi_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { }, }; MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); @@ -50,6 +52,8 @@ MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index e5f2fb884bf3..9974f201a08f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -21,12 +21,19 @@ #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> +#include <linux/gcd.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/pcm_params.h> #include <sound/tlv.h> #include "pcm512x.h" +#define DIV_ROUND_DOWN_ULL(ll, d) \ + ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) +#define DIV_ROUND_CLOSEST_ULL(ll, d) \ + ({ unsigned long long _tmp = (ll)+(d)/2; do_div(_tmp, d); _tmp; }) + #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", @@ -39,6 +46,14 @@ struct pcm512x_priv { struct clk *sclk; struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES]; struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES]; + int fmt; + int pll_in; + int pll_out; + int pll_r; + int pll_j; + int pll_d; + int pll_p; + unsigned long real_pll; }; /* @@ -69,6 +84,7 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_MUTE, 0x00 }, { PCM512x_DSP, 0x00 }, { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_REF, 0x00 }, { PCM512x_DAC_ROUTING, 0x11 }, { PCM512x_DSP_PROGRAM, 0x01 }, { PCM512x_CLKDET, 0x00 }, @@ -87,6 +103,25 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_ANALOG_GAIN_BOOST, 0x00 }, { PCM512x_VCOM_CTRL_1, 0x00 }, { PCM512x_VCOM_CTRL_2, 0x01 }, + { PCM512x_BCLK_LRCLK_CFG, 0x00 }, + { PCM512x_MASTER_MODE, 0x7c }, + { PCM512x_GPIO_DACIN, 0x00 }, + { PCM512x_GPIO_PLLIN, 0x00 }, + { PCM512x_SYNCHRONIZE, 0x10 }, + { PCM512x_PLL_COEFF_0, 0x00 }, + { PCM512x_PLL_COEFF_1, 0x00 }, + { PCM512x_PLL_COEFF_2, 0x00 }, + { PCM512x_PLL_COEFF_3, 0x00 }, + { PCM512x_PLL_COEFF_4, 0x00 }, + { PCM512x_DSP_CLKDIV, 0x00 }, + { PCM512x_DAC_CLKDIV, 0x00 }, + { PCM512x_NCP_CLKDIV, 0x00 }, + { PCM512x_OSR_CLKDIV, 0x00 }, + { PCM512x_MASTER_CLKDIV_1, 0x00 }, + { PCM512x_MASTER_CLKDIV_2, 0x00 }, + { PCM512x_FS_SPEED_MODE, 0x00 }, + { PCM512x_IDAC_1, 0x01 }, + { PCM512x_IDAC_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) @@ -103,6 +138,10 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_DSP_GPIO_INPUT: case PCM512x_MASTER_MODE: case PCM512x_PLL_REF: + case PCM512x_DAC_REF: + case PCM512x_GPIO_DACIN: + case PCM512x_GPIO_PLLIN: + case PCM512x_SYNCHRONIZE: case PCM512x_PLL_COEFF_0: case PCM512x_PLL_COEFF_1: case PCM512x_PLL_COEFF_2: @@ -143,6 +182,7 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_RATE_DET_2: case PCM512x_RATE_DET_3: case PCM512x_RATE_DET_4: + case PCM512x_CLOCK_STATUS: case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: @@ -154,6 +194,8 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_VCOM_CTRL_1: case PCM512x_VCOM_CTRL_2: case PCM512x_CRAM_CTRL: + case PCM512x_FLEX_A: + case PCM512x_FLEX_B: return true; default: /* There are 256 raw register addresses */ @@ -170,6 +212,7 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) case PCM512x_RATE_DET_2: case PCM512x_RATE_DET_3: case PCM512x_RATE_DET_4: + case PCM512x_CLOCK_STATUS: case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: @@ -188,8 +231,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", - "Fixed process flow", "High attenuation with de-emphasis", + "Fixed process flow", "Ringing-less low latency FIR", }; @@ -277,7 +320,7 @@ SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r), SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_ACTL_SHIFT, 1, 0), SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT, - PCM512x_AMLR_SHIFT, 1, 0), + PCM512x_AMRE_SHIFT, 1, 0), SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf), SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds), @@ -303,6 +346,136 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static const u32 pcm512x_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, 384000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_slave = { + .count = ARRAY_SIZE(pcm512x_dai_rates), + .list = pcm512x_dai_rates, +}; + +static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval ranges[2]; + int frame_size; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (frame_size) { + case 32: + /* No hole when the frame size is 32. */ + return 0; + case 48: + case 64: + /* There is only one hole in the range of supported + * rates, but it moves with the frame size. + */ + memset(ranges, 0, sizeof(ranges)); + ranges[0].min = 8000; + ranges[0].max = 25000000 / frame_size / 2; + ranges[1].min = DIV_ROUND_UP(16000000, frame_size); + ranges[1].max = 384000; + break; + default: + return -EINVAL; + } + + return snd_interval_ranges(hw_param_interval(params, rule->var), + ARRAY_SIZE(ranges), ranges, 0); +} + +static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + struct device *dev = dai->dev; + struct snd_pcm_hw_constraint_ratnums *constraints_no_pll; + struct snd_ratnum *rats_no_pll; + + if (IS_ERR(pcm512x->sclk)) { + dev_err(dev, "Need SCLK for master mode: %ld\n", + PTR_ERR(pcm512x->sclk)); + return PTR_ERR(pcm512x->sclk); + } + + if (pcm512x->pll_out) + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + pcm512x_hw_rule_rate, + NULL, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + + constraints_no_pll = devm_kzalloc(dev, sizeof(*constraints_no_pll), + GFP_KERNEL); + if (!constraints_no_pll) + return -ENOMEM; + constraints_no_pll->nrats = 1; + rats_no_pll = devm_kzalloc(dev, sizeof(*rats_no_pll), GFP_KERNEL); + if (!rats_no_pll) + return -ENOMEM; + constraints_no_pll->rats = rats_no_pll; + rats_no_pll->num = clk_get_rate(pcm512x->sclk) / 64; + rats_no_pll->den_min = 1; + rats_no_pll->den_max = 128; + rats_no_pll->den_step = 1; + + return snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + constraints_no_pll); +} + +static int pcm512x_dai_startup_slave(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + struct device *dev = dai->dev; + struct regmap *regmap = pcm512x->regmap; + + if (IS_ERR(pcm512x->sclk)) { + dev_info(dev, "No SCLK, using BCLK: %ld\n", + PTR_ERR(pcm512x->sclk)); + + /* Disable reporting of missing SCLK as an error */ + regmap_update_bits(regmap, PCM512x_ERROR_DETECT, + PCM512x_IDCH, PCM512x_IDCH); + + /* Switch PLL input to BCLK */ + regmap_update_bits(regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF_BCK); + } + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_slave); +} + +static int pcm512x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + return pcm512x_dai_startup_master(substream, dai); + + case SND_SOC_DAIFMT_CBS_CFS: + return pcm512x_dai_startup_slave(substream, dai); + + default: + return -EINVAL; + } +} + static int pcm512x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -340,17 +513,717 @@ static int pcm512x_set_bias_level(struct snd_soc_codec *codec, return 0; } +static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, + unsigned long bclk_rate) +{ + struct device *dev = dai->dev; + unsigned long sck_rate; + int pow2; + + /* 64 MHz <= pll_rate <= 100 MHz, VREF mode */ + /* 16 MHz <= sck_rate <= 25 MHz, VREF mode */ + + /* select sck_rate as a multiple of bclk_rate but still with + * as many factors of 2 as possible, as that makes it easier + * to find a fast DAC rate + */ + pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + for (; pow2; pow2 >>= 1) { + sck_rate = rounddown(25000000, bclk_rate * pow2); + if (sck_rate >= 16000000) + break; + } + if (!pow2) { + dev_err(dev, "Impossible to generate a suitable SCK\n"); + return 0; + } + + dev_dbg(dev, "sck_rate %lu\n", sck_rate); + return sck_rate; +} + +/* pll_rate = pllin_rate * R * J.D / P + * 1 <= R <= 16 + * 1 <= J <= 63 + * 0 <= D <= 9999 + * 1 <= P <= 15 + * 64 MHz <= pll_rate <= 100 MHz + * if D == 0 + * 1 MHz <= pllin_rate / P <= 20 MHz + * else if D > 0 + * 6.667 MHz <= pllin_rate / P <= 20 MHz + * 4 <= J <= 11 + * R = 1 + */ +static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai, + unsigned long pllin_rate, + unsigned long pll_rate) +{ + struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + unsigned long common; + int R, J, D, P; + unsigned long K; /* 10000 * J.D */ + unsigned long num; + unsigned long den; + + common = gcd(pll_rate, pllin_rate); + dev_dbg(dev, "pll %lu pllin %lu common %lu\n", + pll_rate, pllin_rate, common); + num = pll_rate / common; + den = pllin_rate / common; + + /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */ + if (pllin_rate / den > 20000000 && num < 8) { + num *= 20000000 / (pllin_rate / den); + den *= 20000000 / (pllin_rate / den); + } + dev_dbg(dev, "num / den = %lu / %lu\n", num, den); + + P = den; + if (den <= 15 && num <= 16 * 63 + && 1000000 <= pllin_rate / P && pllin_rate / P <= 20000000) { + /* Try the case with D = 0 */ + D = 0; + /* factor 'num' into J and R, such that R <= 16 and J <= 63 */ + for (R = 16; R; R--) { + if (num % R) + continue; + J = num / R; + if (J == 0 || J > 63) + continue; + + dev_dbg(dev, "R * J / P = %d * %d / %d\n", R, J, P); + pcm512x->real_pll = pll_rate; + goto done; + } + /* no luck */ + } + + R = 1; + + if (num > 0xffffffffUL / 10000) + goto fallback; + + /* Try to find an exact pll_rate using the D > 0 case */ + common = gcd(10000 * num, den); + num = 10000 * num / common; + den /= common; + dev_dbg(dev, "num %lu den %lu common %lu\n", num, den, common); + + for (P = den; P <= 15; P++) { + if (pllin_rate / P < 6667000 || 200000000 < pllin_rate / P) + continue; + if (num * P % den) + continue; + K = num * P / den; + /* J == 12 is ok if D == 0 */ + if (K < 40000 || K > 120000) + continue; + + J = K / 10000; + D = K % 10000; + dev_dbg(dev, "J.D / P = %d.%04d / %d\n", J, D, P); + pcm512x->real_pll = pll_rate; + goto done; + } + + /* Fall back to an approximate pll_rate */ + +fallback: + /* find smallest possible P */ + P = DIV_ROUND_UP(pllin_rate, 20000000); + if (!P) + P = 1; + else if (P > 15) { + dev_err(dev, "Need a slower clock as pll-input\n"); + return -EINVAL; + } + if (pllin_rate / P < 6667000) { + dev_err(dev, "Need a faster clock as pll-input\n"); + return -EINVAL; + } + K = DIV_ROUND_CLOSEST_ULL(10000ULL * pll_rate * P, pllin_rate); + if (K < 40000) + K = 40000; + /* J == 12 is ok if D == 0 */ + if (K > 120000) + K = 120000; + J = K / 10000; + D = K % 10000; + dev_dbg(dev, "J.D / P ~ %d.%04d / %d\n", J, D, P); + pcm512x->real_pll = DIV_ROUND_DOWN_ULL((u64)K * pllin_rate, 10000 * P); + +done: + pcm512x->pll_r = R; + pcm512x->pll_j = J; + pcm512x->pll_d = D; + pcm512x->pll_p = P; + return 0; +} + +static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, + unsigned long osr_rate, + unsigned long pllin_rate) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + unsigned long dac_rate; + + if (!pcm512x->pll_out) + return 0; /* no PLL to bypass, force SCK as DAC input */ + + if (pllin_rate % osr_rate) + return 0; /* futile, quit early */ + + /* run DAC no faster than 6144000 Hz */ + for (dac_rate = rounddown(6144000, osr_rate); + dac_rate; + dac_rate -= osr_rate) { + + if (pllin_rate / dac_rate > 128) + return 0; /* DAC divider would be too big */ + + if (!(pllin_rate % dac_rate)) + return dac_rate; + + dac_rate -= osr_rate; + } + + return 0; +} + +static int pcm512x_set_dividers(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + unsigned long pllin_rate = 0; + unsigned long pll_rate; + unsigned long sck_rate; + unsigned long mck_rate; + unsigned long bclk_rate; + unsigned long sample_rate; + unsigned long osr_rate; + unsigned long dacsrc_rate; + int bclk_div; + int lrclk_div; + int dsp_div; + int dac_div; + unsigned long dac_rate; + int ncp_div; + int osr_div; + int ret; + int idac; + int fssp; + int gpio; + + lrclk_div = snd_soc_params_to_frame_size(params); + if (lrclk_div == 0) { + dev_err(dev, "No LRCLK?\n"); + return -EINVAL; + } + + if (!pcm512x->pll_out) { + sck_rate = clk_get_rate(pcm512x->sclk); + bclk_div = params->rate_den * 64 / lrclk_div; + bclk_rate = DIV_ROUND_CLOSEST(sck_rate, bclk_div); + + mck_rate = sck_rate; + } else { + ret = snd_soc_params_to_bclk(params); + if (ret < 0) { + dev_err(dev, "Failed to find suitable BCLK: %d\n", ret); + return ret; + } + if (ret == 0) { + dev_err(dev, "No BCLK?\n"); + return -EINVAL; + } + bclk_rate = ret; + + pllin_rate = clk_get_rate(pcm512x->sclk); + + sck_rate = pcm512x_find_sck(dai, bclk_rate); + if (!sck_rate) + return -EINVAL; + pll_rate = 4 * sck_rate; + + ret = pcm512x_find_pll_coeff(dai, pllin_rate, pll_rate); + if (ret != 0) + return ret; + + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_0, pcm512x->pll_p - 1); + if (ret != 0) { + dev_err(dev, "Failed to write PLL P: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_1, pcm512x->pll_j); + if (ret != 0) { + dev_err(dev, "Failed to write PLL J: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_2, pcm512x->pll_d >> 8); + if (ret != 0) { + dev_err(dev, "Failed to write PLL D msb: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_3, pcm512x->pll_d & 0xff); + if (ret != 0) { + dev_err(dev, "Failed to write PLL D lsb: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_4, pcm512x->pll_r - 1); + if (ret != 0) { + dev_err(dev, "Failed to write PLL R: %d\n", ret); + return ret; + } + + mck_rate = pcm512x->real_pll; + + bclk_div = DIV_ROUND_CLOSEST(sck_rate, bclk_rate); + } + + if (bclk_div > 128) { + dev_err(dev, "Failed to find BCLK divider\n"); + return -EINVAL; + } + + /* the actual rate */ + sample_rate = sck_rate / bclk_div / lrclk_div; + osr_rate = 16 * sample_rate; + + /* run DSP no faster than 50 MHz */ + dsp_div = mck_rate > 50000000 ? 2 : 1; + + dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); + if (dac_rate) { + /* the desired clock rate is "compatible" with the pll input + * clock, so use that clock as dac input instead of the pll + * output clock since the pll will introduce jitter and thus + * noise. + */ + dev_dbg(dev, "using pll input as dac input\n"); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_DAC_REF, + PCM512x_SDAC, PCM512x_SDAC_GPIO); + if (ret != 0) { + dev_err(codec->dev, + "Failed to set gpio as dacref: %d\n", ret); + return ret; + } + + gpio = PCM512x_GREF_GPIO1 + pcm512x->pll_in - 1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_DACIN, + PCM512x_GREF, gpio); + if (ret != 0) { + dev_err(codec->dev, + "Failed to set gpio %d as dacin: %d\n", + pcm512x->pll_in, ret); + return ret; + } + + dacsrc_rate = pllin_rate; + } else { + /* run DAC no faster than 6144000 Hz */ + unsigned long dac_mul = 6144000 / osr_rate; + unsigned long sck_mul = sck_rate / osr_rate; + + for (; dac_mul; dac_mul--) { + if (!(sck_mul % dac_mul)) + break; + } + if (!dac_mul) { + dev_err(dev, "Failed to find DAC rate\n"); + return -EINVAL; + } + + dac_rate = dac_mul * osr_rate; + dev_dbg(dev, "dac_rate %lu sample_rate %lu\n", + dac_rate, sample_rate); + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_DAC_REF, + PCM512x_SDAC, PCM512x_SDAC_SCK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to set sck as dacref: %d\n", ret); + return ret; + } + + dacsrc_rate = sck_rate; + } + + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); + if (dac_div > 128) { + dev_err(dev, "Failed to find DAC divider\n"); + return -EINVAL; + } + + ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); + if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + /* run NCP no faster than 2048000 Hz, but why? */ + ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + if (ncp_div > 128) { + dev_err(dev, "Failed to find NCP divider\n"); + return -EINVAL; + } + } + + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + + idac = mck_rate / (dsp_div * sample_rate); + + ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write DSP divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_DAC_CLKDIV, dac_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write DAC divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_NCP_CLKDIV, ncp_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write NCP divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_OSR_CLKDIV, osr_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write OSR divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_MASTER_CLKDIV_1, bclk_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write BCLK divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, + PCM512x_MASTER_CLKDIV_2, lrclk_div - 1); + if (ret != 0) { + dev_err(dev, "Failed to write LRCLK divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_IDAC_1, idac >> 8); + if (ret != 0) { + dev_err(dev, "Failed to write IDAC msb divider: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_IDAC_2, idac & 0xff); + if (ret != 0) { + dev_err(dev, "Failed to write IDAC lsb divider: %d\n", ret); + return ret; + } + + if (sample_rate <= 48000) + fssp = PCM512x_FSSP_48KHZ; + else if (sample_rate <= 96000) + fssp = PCM512x_FSSP_96KHZ; + else if (sample_rate <= 192000) + fssp = PCM512x_FSSP_192KHZ; + else + fssp = PCM512x_FSSP_384KHZ; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_FS_SPEED_MODE, + PCM512x_FSSP, fssp); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fs speed: %d\n", ret); + return ret; + } + + dev_dbg(codec->dev, "DSP divider %d\n", dsp_div); + dev_dbg(codec->dev, "DAC divider %d\n", dac_div); + dev_dbg(codec->dev, "NCP divider %d\n", ncp_div); + dev_dbg(codec->dev, "OSR divider %d\n", osr_div); + dev_dbg(codec->dev, "BCK divider %d\n", bclk_div); + dev_dbg(codec->dev, "LRCK divider %d\n", lrclk_div); + dev_dbg(codec->dev, "IDAC %d\n", idac); + dev_dbg(codec->dev, "1<<FSSP %d\n", 1 << fssp); + + return 0; +} + +static int pcm512x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + int alen; + int gpio; + int clock_output; + int master_mode; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + switch (snd_pcm_format_width(params_format(params))) { + case 16: + alen = PCM512x_ALEN_16; + break; + case 20: + alen = PCM512x_ALEN_20; + break; + case 24: + alen = PCM512x_ALEN_24; + break; + case 32: + alen = PCM512x_ALEN_32; + break; + default: + dev_err(codec->dev, "Bad frame size: %d\n", + snd_pcm_format_width(params_format(params))); + return -EINVAL; + } + + switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ret = regmap_update_bits(pcm512x->regmap, + PCM512x_BCLK_LRCLK_CFG, + PCM512x_BCKP + | PCM512x_BCKO | PCM512x_LRKO, + 0); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable slave mode: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT, + PCM512x_DCAS, 0); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable clock divider autoset: %d\n", + ret); + return ret; + } + return 0; + case SND_SOC_DAIFMT_CBM_CFM: + clock_output = PCM512x_BCKO | PCM512x_LRKO; + master_mode = PCM512x_RLRK | PCM512x_RBCK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + clock_output = PCM512x_BCKO; + master_mode = PCM512x_RBCK; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_I2S_1, + PCM512x_ALEN, alen); + if (ret != 0) { + dev_err(codec->dev, "Failed to set frame size: %d\n", ret); + return ret; + } + + if (pcm512x->pll_out) { + ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_A, 0x11); + if (ret != 0) { + dev_err(codec->dev, "Failed to set FLEX_A: %d\n", ret); + return ret; + } + + ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_B, 0xff); + if (ret != 0) { + dev_err(codec->dev, "Failed to set FLEX_B: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT, + PCM512x_IDFS | PCM512x_IDBK + | PCM512x_IDSK | PCM512x_IDCH + | PCM512x_IDCM | PCM512x_DCAS + | PCM512x_IPLK, + PCM512x_IDFS | PCM512x_IDBK + | PCM512x_IDSK | PCM512x_IDCH + | PCM512x_DCAS); + if (ret != 0) { + dev_err(codec->dev, + "Failed to ignore auto-clock failures: %d\n", + ret); + return ret; + } + } else { + ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT, + PCM512x_IDFS | PCM512x_IDBK + | PCM512x_IDSK | PCM512x_IDCH + | PCM512x_IDCM | PCM512x_DCAS + | PCM512x_IPLK, + PCM512x_IDFS | PCM512x_IDBK + | PCM512x_IDSK | PCM512x_IDCH + | PCM512x_DCAS | PCM512x_IPLK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to ignore auto-clock failures: %d\n", + ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN, + PCM512x_PLLE, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to disable pll: %d\n", ret); + return ret; + } + } + + ret = pcm512x_set_dividers(dai, params); + if (ret != 0) + return ret; + + if (pcm512x->pll_out) { + ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF_GPIO); + if (ret != 0) { + dev_err(codec->dev, + "Failed to set gpio as pllref: %d\n", ret); + return ret; + } + + gpio = PCM512x_GREF_GPIO1 + pcm512x->pll_in - 1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_PLLIN, + PCM512x_GREF, gpio); + if (ret != 0) { + dev_err(codec->dev, + "Failed to set gpio %d as pllin: %d\n", + pcm512x->pll_in, ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN, + PCM512x_PLLE, PCM512x_PLLE); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable pll: %d\n", ret); + return ret; + } + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_BCLK_LRCLK_CFG, + PCM512x_BCKP | PCM512x_BCKO | PCM512x_LRKO, + clock_output); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable clock output: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_MASTER_MODE, + PCM512x_RLRK | PCM512x_RBCK, + master_mode); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable master mode: %d\n", ret); + return ret; + } + + if (pcm512x->pll_out) { + gpio = PCM512x_G1OE << (pcm512x->pll_out - 1); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, + gpio, gpio); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable gpio %d: %d\n", + pcm512x->pll_out, ret); + return ret; + } + + gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_out - 1; + ret = regmap_update_bits(pcm512x->regmap, gpio, + PCM512x_GxSL, PCM512x_GxSL_PLLCK); + if (ret != 0) { + dev_err(codec->dev, "Failed to output pll on %d: %d\n", + ret, pcm512x->pll_out); + return ret; + } + + gpio = PCM512x_G1OE << (4 - 1); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, + gpio, gpio); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable gpio %d: %d\n", + 4, ret); + return ret; + } + + gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; + ret = regmap_update_bits(pcm512x->regmap, gpio, + PCM512x_GxSL, PCM512x_GxSL_PLLLK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to output pll lock on %d: %d\n", + ret, 4); + return ret; + } + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, + PCM512x_RQSY, PCM512x_RQSY_HALT); + if (ret != 0) { + dev_err(codec->dev, "Failed to halt clocks: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, + PCM512x_RQSY, PCM512x_RQSY_RESUME); + if (ret != 0) { + dev_err(codec->dev, "Failed to resume clocks: %d\n", ret); + return ret; + } + + return 0; +} + +static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + pcm512x->fmt = fmt; + + return 0; +} + +static const struct snd_soc_dai_ops pcm512x_dai_ops = { + .startup = pcm512x_dai_startup, + .hw_params = pcm512x_hw_params, + .set_fmt = pcm512x_set_fmt, +}; + static struct snd_soc_dai_driver pcm512x_dai = { .name = "pcm512x-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, + .ops = &pcm512x_dai_ops, }; static struct snd_soc_codec_driver pcm512x_codec_driver = { @@ -448,21 +1321,9 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) } pcm512x->sclk = devm_clk_get(dev, NULL); - if (IS_ERR(pcm512x->sclk)) { - if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; - - dev_info(dev, "No SCLK, using BCLK: %ld\n", - PTR_ERR(pcm512x->sclk)); - - /* Disable reporting of missing SCLK as an error */ - regmap_update_bits(regmap, PCM512x_ERROR_DETECT, - PCM512x_IDCH, PCM512x_IDCH); - - /* Switch PLL input to BCLK */ - regmap_update_bits(regmap, PCM512x_PLL_REF, - PCM512x_SREF, PCM512x_SREF); - } else { + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (!IS_ERR(pcm512x->sclk)) { ret = clk_prepare_enable(pcm512x->sclk); if (ret != 0) { dev_err(dev, "Failed to enable SCLK: %d\n", ret); @@ -483,6 +1344,43 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) pm_runtime_enable(dev); pm_runtime_idle(dev); +#ifdef CONFIG_OF + if (dev->of_node) { + const struct device_node *np = dev->of_node; + u32 val; + + if (of_property_read_u32(np, "pll-in", &val) >= 0) { + if (val > 6) { + dev_err(dev, "Invalid pll-in\n"); + ret = -EINVAL; + goto err_clk; + } + pcm512x->pll_in = val; + } + + if (of_property_read_u32(np, "pll-out", &val) >= 0) { + if (val > 6) { + dev_err(dev, "Invalid pll-out\n"); + ret = -EINVAL; + goto err_clk; + } + pcm512x->pll_out = val; + } + + if (!pcm512x->pll_in != !pcm512x->pll_out) { + dev_err(dev, + "Error: both pll-in and pll-out, or none\n"); + ret = -EINVAL; + goto err_clk; + } + if (pcm512x->pll_in && pcm512x->pll_in == pcm512x->pll_out) { + dev_err(dev, "Error: pll-in == pll-out\n"); + ret = -EINVAL; + goto err_clk; + } + } +#endif + ret = snd_soc_register_codec(dev, &pcm512x_codec_driver, &pcm512x_dai, 1); if (ret != 0) { diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index 6ee76aaca09a..b7c310207223 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -37,6 +37,10 @@ #define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10) #define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12) #define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13) +#define PCM512x_DAC_REF (PCM512x_PAGE_BASE(0) + 14) +#define PCM512x_GPIO_DACIN (PCM512x_PAGE_BASE(0) + 16) +#define PCM512x_GPIO_PLLIN (PCM512x_PAGE_BASE(0) + 18) +#define PCM512x_SYNCHRONIZE (PCM512x_PAGE_BASE(0) + 19) #define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20) #define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21) #define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22) @@ -77,6 +81,7 @@ #define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92) #define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93) #define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94) +#define PCM512x_CLOCK_STATUS (PCM512x_PAGE_BASE(0) + 95) #define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108) #define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119) #define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120) @@ -91,7 +96,10 @@ #define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1) -#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1) +#define PCM512x_FLEX_A (PCM512x_PAGE_BASE(253) + 63) +#define PCM512x_FLEX_B (PCM512x_PAGE_BASE(253) + 64) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(253) + 64) /* Page 0, Register 1 - reset */ #define PCM512x_RSTR (1 << 0) @@ -108,8 +116,8 @@ #define PCM512x_RQML_SHIFT 4 /* Page 0, Register 4 - PLL */ -#define PCM512x_PLCE (1 << 0) -#define PCM512x_RLCE_SHIFT 0 +#define PCM512x_PLLE (1 << 0) +#define PCM512x_PLLE_SHIFT 0 #define PCM512x_PLCK (1 << 4) #define PCM512x_PLCK_SHIFT 4 @@ -119,8 +127,66 @@ #define PCM512x_DEMP (1 << 4) #define PCM512x_DEMP_SHIFT 4 +/* Page 0, Register 8 - GPIO output enable */ +#define PCM512x_G1OE (1 << 0) +#define PCM512x_G2OE (1 << 1) +#define PCM512x_G3OE (1 << 2) +#define PCM512x_G4OE (1 << 3) +#define PCM512x_G5OE (1 << 4) +#define PCM512x_G6OE (1 << 5) + +/* Page 0, Register 9 - BCK, LRCLK configuration */ +#define PCM512x_LRKO (1 << 0) +#define PCM512x_LRKO_SHIFT 0 +#define PCM512x_BCKO (1 << 4) +#define PCM512x_BCKO_SHIFT 4 +#define PCM512x_BCKP (1 << 5) +#define PCM512x_BCKP_SHIFT 5 + +/* Page 0, Register 12 - Master mode BCK, LRCLK reset */ +#define PCM512x_RLRK (1 << 0) +#define PCM512x_RLRK_SHIFT 0 +#define PCM512x_RBCK (1 << 1) +#define PCM512x_RBCK_SHIFT 1 + /* Page 0, Register 13 - PLL reference */ -#define PCM512x_SREF (1 << 4) +#define PCM512x_SREF (7 << 4) +#define PCM512x_SREF_SHIFT 4 +#define PCM512x_SREF_SCK (0 << 4) +#define PCM512x_SREF_BCK (1 << 4) +#define PCM512x_SREF_GPIO (3 << 4) + +/* Page 0, Register 14 - DAC reference */ +#define PCM512x_SDAC (7 << 4) +#define PCM512x_SDAC_SHIFT 4 +#define PCM512x_SDAC_MCK (0 << 4) +#define PCM512x_SDAC_PLL (1 << 4) +#define PCM512x_SDAC_SCK (3 << 4) +#define PCM512x_SDAC_BCK (4 << 4) +#define PCM512x_SDAC_GPIO (5 << 4) + +/* Page 0, Register 16, 18 - GPIO source for DAC, PLL */ +#define PCM512x_GREF (7 << 0) +#define PCM512x_GREF_SHIFT 0 +#define PCM512x_GREF_GPIO1 (0 << 0) +#define PCM512x_GREF_GPIO2 (1 << 0) +#define PCM512x_GREF_GPIO3 (2 << 0) +#define PCM512x_GREF_GPIO4 (3 << 0) +#define PCM512x_GREF_GPIO5 (4 << 0) +#define PCM512x_GREF_GPIO6 (5 << 0) + +/* Page 0, Register 19 - synchronize */ +#define PCM512x_RQSY (1 << 0) +#define PCM512x_RQSY_RESUME (0 << 0) +#define PCM512x_RQSY_HALT (1 << 0) + +/* Page 0, Register 34 - fs speed mode */ +#define PCM512x_FSSP (3 << 0) +#define PCM512x_FSSP_SHIFT 0 +#define PCM512x_FSSP_48KHZ (0 << 0) +#define PCM512x_FSSP_96KHZ (1 << 0) +#define PCM512x_FSSP_192KHZ (2 << 0) +#define PCM512x_FSSP_384KHZ (3 << 0) /* Page 0, Register 37 - Error detection */ #define PCM512x_IPLK (1 << 0) @@ -131,6 +197,20 @@ #define PCM512x_IDBK (1 << 5) #define PCM512x_IDFS (1 << 6) +/* Page 0, Register 40 - I2S configuration */ +#define PCM512x_ALEN (3 << 0) +#define PCM512x_ALEN_SHIFT 0 +#define PCM512x_ALEN_16 (0 << 0) +#define PCM512x_ALEN_20 (1 << 0) +#define PCM512x_ALEN_24 (2 << 0) +#define PCM512x_ALEN_32 (3 << 0) +#define PCM512x_AFMT (3 << 4) +#define PCM512x_AFMT_SHIFT 4 +#define PCM512x_AFMT_I2S (0 << 4) +#define PCM512x_AFMT_DSP (1 << 4) +#define PCM512x_AFMT_RTJ (2 << 4) +#define PCM512x_AFMT_LTJ (3 << 4) + /* Page 0, Register 42 - DAC routing */ #define PCM512x_AUPR_SHIFT 0 #define PCM512x_AUPL_SHIFT 4 @@ -152,7 +232,26 @@ /* Page 0, Register 65 - Digital mute enables */ #define PCM512x_ACTL_SHIFT 2 #define PCM512x_AMLE_SHIFT 1 -#define PCM512x_AMLR_SHIFT 0 +#define PCM512x_AMRE_SHIFT 0 + +/* Page 0, Register 80-85, GPIO output selection */ +#define PCM512x_GxSL (31 << 0) +#define PCM512x_GxSL_SHIFT 0 +#define PCM512x_GxSL_OFF (0 << 0) +#define PCM512x_GxSL_DSP (1 << 0) +#define PCM512x_GxSL_REG (2 << 0) +#define PCM512x_GxSL_AMUTB (3 << 0) +#define PCM512x_GxSL_AMUTL (4 << 0) +#define PCM512x_GxSL_AMUTR (5 << 0) +#define PCM512x_GxSL_CLKI (6 << 0) +#define PCM512x_GxSL_SDOUT (7 << 0) +#define PCM512x_GxSL_ANMUL (8 << 0) +#define PCM512x_GxSL_ANMUR (9 << 0) +#define PCM512x_GxSL_PLLLK (10 << 0) +#define PCM512x_GxSL_CPCLK (11 << 0) +#define PCM512x_GxSL_UV0_7 (14 << 0) +#define PCM512x_GxSL_UV0_3 (15 << 0) +#define PCM512x_GxSL_PLLCK (16 << 0) /* Page 1, Register 2 - analog volume control */ #define PCM512x_RAGN_SHIFT 0 diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe463102..d0698891b69e 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -305,6 +305,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *hp = false; *mic = false; + if (!rt286->codec) + return -EINVAL; if (rt286->pdata.cbj_en) { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; @@ -403,7 +405,8 @@ EXPORT_SYMBOL_GPL(rt286_mic_detect); static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); if (rt286->clk_id == RT286_SCLK_S_MCLK) return 1; @@ -417,6 +420,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static const struct snd_kcontrol_new rt286_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 7, 1, 1), SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, @@ -500,7 +505,7 @@ SOC_DAPM_ENUM("SPO source", rt286_spo_enum); static int rt286_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -522,7 +527,7 @@ static int rt286_spk_event(struct snd_soc_dapm_widget *w, static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -538,36 +543,10 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_adc_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int nid; - - nid = (w->reg >> 20) & 0xff; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7000); - break; - case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7080); - break; - default: - return 0; - } - - return 0; -} - static int rt286_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -585,7 +564,7 @@ static int rt286_vref_event(struct snd_soc_dapm_widget *w, static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -604,7 +583,7 @@ static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, static int rt286_mic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -667,12 +646,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), /* ADC Mux */ - SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, - &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, - &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux), /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), @@ -861,10 +838,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, RT286_I2S_CTRL1, 0x0018, d_len_code << 3); dev_dbg(codec->dev, "format val = 0x%x\n", val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); - else - snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); return 0; } diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 6d7b7ca7d530..c61852742ee3 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -287,70 +287,78 @@ static const struct snd_kcontrol_new rt5631_snd_controls[] = { static int check_sysclk1_source(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_GLOBAL_CLK_CTRL); + reg = snd_soc_read(codec, RT5631_GLOBAL_CLK_CTRL); return reg & RT5631_SYSCLK_SOUR_SEL_PLL; } static int check_dmic_used(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(source->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); return rt5631->dmic_used_flag; } static int check_dacl_to_outmixl(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_OUTMIXER_L_CTRL); + reg = snd_soc_read(codec, RT5631_OUTMIXER_L_CTRL); return !(reg & RT5631_M_DAC_L_TO_OUTMIXER_L); } static int check_dacr_to_outmixr(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_OUTMIXER_R_CTRL); + reg = snd_soc_read(codec, RT5631_OUTMIXER_R_CTRL); return !(reg & RT5631_M_DAC_R_TO_OUTMIXER_R); } static int check_dacl_to_spkmixl(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + reg = snd_soc_read(codec, RT5631_SPK_MIXER_CTRL); return !(reg & RT5631_M_DAC_L_TO_SPKMIXER_L); } static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_SPK_MIXER_CTRL); + reg = snd_soc_read(codec, RT5631_SPK_MIXER_CTRL); return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R); } static int check_adcl_select(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + reg = snd_soc_read(codec, RT5631_ADC_REC_MIXER); return !(reg & RT5631_M_MIC1_TO_RECMIXER_L); } static int check_adcr_select(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; - reg = snd_soc_read(source->codec, RT5631_ADC_REC_MIXER); + reg = snd_soc_read(codec, RT5631_ADC_REC_MIXER); return !(reg & RT5631_M_MIC2_TO_RECMIXER_R); } @@ -556,7 +564,7 @@ static void depop_seq_mute_stage(struct snd_soc_codec *codec, int enable) static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -590,7 +598,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, static int set_dmic_params(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); switch (rt5631->rx_rate) { diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c3f2decd643c..178e55d4d481 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -458,7 +458,7 @@ static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); int idx = -EINVAL; @@ -475,9 +475,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int val; - val = snd_soc_read(source->codec, RT5640_GLB_CLK); + val = snd_soc_read(codec, RT5640_GLB_CLK); val &= RT5640_SCLK_SRC_MASK; if (val == RT5640_SCLK_SRC_PLL1) return 1; @@ -963,7 +964,7 @@ static void rt5640_pmu_depop(struct snd_soc_codec *codec) static int rt5640_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -987,7 +988,7 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1003,7 +1004,7 @@ static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -2124,6 +2125,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 27141e2df878..c9a4c5be083b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -31,6 +31,7 @@ #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 +#define RT5650_DEVICE_ID 0x6419 #define RT5645_PR_RANGE_BASE (0xff + 1) #define RT5645_PR_SPACING 0x100 @@ -59,6 +60,10 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) +static const struct reg_default rt5650_init_list[] = { + {0xf6, 0x0100}, +}; + static const struct reg_default rt5645_reg[] = { { 0x00, 0x0000 }, { 0x01, 0xc8c8 }, @@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = { { 0x2a, 0x5656 }, { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, { 0x2f, 0x1002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, @@ -193,6 +199,8 @@ static const struct reg_default rt5645_reg[] = { { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, { 0xdd, 0x001b }, + { 0xdf, 0x0008 }, + { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, { 0xe7, 0x0200 }, { 0xec, 0xb300 }, @@ -242,6 +250,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) case RT5645_IRQ_CTRL3: case RT5645_INT_IRQ_ST: case RT5645_IL_CMD: + case RT5650_4BTN_IL_CMD1: case RT5645_VENDOR_ID: case RT5645_VENDOR_ID1: case RT5645_VENDOR_ID2: @@ -287,6 +296,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_STO_DAC_MIXER: case RT5645_MONO_DAC_MIXER: case RT5645_DIG_MIXER: + case RT5650_A_DAC_SOUR: case RT5645_DIG_INF1_DATA: case RT5645_PDM_OUT_CTRL: case RT5645_REC_L1_MIXER: @@ -378,6 +388,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_IL_CMD: case RT5645_IL_CMD2: case RT5645_IL_CMD3: + case RT5650_4BTN_IL_CMD1: + case RT5650_4BTN_IL_CMD2: case RT5645_DRC1_HL_CTRL1: case RT5645_DRC2_HL_CTRL1: case RT5645_ADC_MONO_HP_CTRL1: @@ -527,7 +539,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); int idx = -EINVAL; @@ -544,9 +556,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int val; - val = snd_soc_read(source->codec, RT5645_GLB_CLK); + val = snd_soc_read(codec, RT5645_GLB_CLK); val &= RT5645_SCLK_SRC_MASK; if (val == RT5645_SCLK_SRC_PLL1) return 1; @@ -557,6 +570,7 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg, shift, val; switch (source->shift) { @@ -588,7 +602,7 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, return 0; } - val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + val = (snd_soc_read(codec, reg) >> shift) & 0xf; switch (val) { case 1: case 2: @@ -601,6 +615,87 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +/** + * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5645 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5645_CLK_SEL_SYS: + case RT5645_CLK_SEL_I2S1_ASRC: + case RT5645_CLK_SEL_I2S2_ASRC: + case RT5645_CLK_SEL_SYS2: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5645_DA_STEREO_FILTER) { + asrc2_mask |= RT5645_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5645_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_DA_MONO_L_FILTER) { + asrc2_mask |= RT5645_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5645_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_DA_MONO_R_FILTER) { + asrc2_mask |= RT5645_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5645_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_STEREO_FILTER) { + asrc2_mask |= RT5645_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5645_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_MONO_L_FILTER) { + asrc3_mask |= RT5645_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5645_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5645_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_MONO_R_FILTER) { + asrc3_mask |= RT5645_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5645_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5645_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5645_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5645_ASRC_3, + asrc3_mask, asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5645_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1007,6 +1102,44 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); +/* MX-2d [3] [2] */ +static const char * const rt5650_a_dac1_src[] = { + "DAC1", "Stereo DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_l_mux = + SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_r_mux = + SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum); + +/* MX-2d [1] [0] */ +static const char * const rt5650_a_dac2_src[] = { + "Stereo DAC Mixer", "Mono DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_l_mux = + SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_r_mux = + SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum); + /* MX-2F [13:12] */ static const char * const rt5645_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "VAD_ADC" @@ -1144,18 +1277,23 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) static int rt5645_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: hp_amp_power(codec, 1); /* headphone unmute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | - RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1175,12 +1313,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: /* headphone mute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, - RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | - RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1205,7 +1347,7 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1232,7 +1374,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, static int rt5645_lout_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1262,7 +1404,7 @@ static int rt5645_lout_event(struct snd_soc_dapm_widget *w, static int rt5645_bst2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1574,6 +1716,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_l_mux), + SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_r_mux), + SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_l_mux), + SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_r_mux), +}; + static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1779,13 +1932,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, - { "DAC L1", NULL, "Stereo DAC MIXL" }, { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R1", NULL, "Stereo DAC MIXR" }, { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC L2", NULL, "Mono DAC MIXL" }, { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R2", NULL, "Mono DAC MIXR" }, { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, { "SPK MIXL", "BST1 Switch", "BST1" }, @@ -1874,6 +2023,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPOR", NULL, "SPK amp" }, }; +static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = { + { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"}, + { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"}, + { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + + { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"}, + { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"}, + + { "DAC L1", NULL, "A DAC1 L Mux" }, + { "DAC R1", NULL, "A DAC1 R Mux" }, + { "DAC L2", NULL, "A DAC2 L Mux" }, + { "DAC R2", NULL, "A DAC2 R Mux" }, +}; + +static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC R2", NULL, "Mono DAC MIXR" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2293,6 +2466,22 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645->codec = codec; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5645: + snd_soc_dapm_add_routes(&codec->dapm, + rt5645_specific_dapm_routes, + ARRAY_SIZE(rt5645_specific_dapm_routes)); + break; + case CODEC_TYPE_RT5650: + snd_soc_dapm_new_controls(&codec->dapm, + rt5650_specific_dapm_widgets, + ARRAY_SIZE(rt5650_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5650_specific_dapm_routes, + ARRAY_SIZE(rt5650_specific_dapm_routes)); + break; + } + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); @@ -2424,6 +2613,7 @@ static const struct regmap_config rt5645_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5650", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); @@ -2456,9 +2646,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); - if (val != RT5645_DEVICE_ID) { + + switch (val) { + case RT5645_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5645; + break; + case RT5650_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5650; + break; + default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645\n", val); + "Device with ID register %x is not rt5645 or rt5650\n", + val); return -ENODEV; } @@ -2469,6 +2668,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ARRAY_SIZE(rt5650_init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", + ret); + } + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index a815e36a2bdb..dbfd98c22f4d 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -47,6 +47,7 @@ #define RT5645_STO_DAC_MIXER 0x2a #define RT5645_MONO_DAC_MIXER 0x2b #define RT5645_DIG_MIXER 0x2c +#define RT5650_A_DAC_SOUR 0x2d #define RT5645_DIG_INF1_DATA 0x2f /* Mixer - PDM */ #define RT5645_PDM_OUT_CTRL 0x31 @@ -150,6 +151,8 @@ #define RT5645_IL_CMD 0xdb #define RT5645_IL_CMD2 0xdc #define RT5645_IL_CMD3 0xdd +#define RT5650_4BTN_IL_CMD1 0xdf +#define RT5650_4BTN_IL_CMD2 0xe0 #define RT5645_DRC1_HL_CTRL1 0xe7 #define RT5645_DRC2_HL_CTRL1 0xe9 #define RT5645_MUTI_DRC_CTRL1 0xea @@ -472,6 +475,12 @@ #define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) #define RT5645_DAC_L2_DAC_R_VOL_SFT 4 +/* Analog DAC1/2 Input Source Control (0x2d) */ +#define RT5650_A_DAC1_L_IN_SFT 3 +#define RT5650_A_DAC1_R_IN_SFT 2 +#define RT5650_A_DAC2_L_IN_SFT 1 +#define RT5650_A_DAC2_R_IN_SFT 0 + /* Digital Interface Data Control (0x2f) */ #define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) #define RT5645_IF1_ADC2_IN_SFT 15 @@ -1111,50 +1120,27 @@ #define RT5645_DMIC_2_M_NOR (0x0 << 8) #define RT5645_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5645_CLK_SEL_SYS (0x0) +#define RT5645_CLK_SEL_I2S1_ASRC (0x1) +#define RT5645_CLK_SEL_I2S2_ASRC (0x2) +#define RT5645_CLK_SEL_SYS2 (0x5) + /* ASRC Control 2 (0x84) */ -#define RT5645_MDA_L_M_MASK (0x1 << 15) -#define RT5645_MDA_L_M_SFT 15 -#define RT5645_MDA_L_M_NOR (0x0 << 15) -#define RT5645_MDA_L_M_ASYN (0x1 << 15) -#define RT5645_MDA_R_M_MASK (0x1 << 14) -#define RT5645_MDA_R_M_SFT 14 -#define RT5645_MDA_R_M_NOR (0x0 << 14) -#define RT5645_MDA_R_M_ASYN (0x1 << 14) -#define RT5645_MAD_L_M_MASK (0x1 << 13) -#define RT5645_MAD_L_M_SFT 13 -#define RT5645_MAD_L_M_NOR (0x0 << 13) -#define RT5645_MAD_L_M_ASYN (0x1 << 13) -#define RT5645_MAD_R_M_MASK (0x1 << 12) -#define RT5645_MAD_R_M_SFT 12 -#define RT5645_MAD_R_M_NOR (0x0 << 12) -#define RT5645_MAD_R_M_ASYN (0x1 << 12) -#define RT5645_ADC_M_MASK (0x1 << 11) -#define RT5645_ADC_M_SFT 11 -#define RT5645_ADC_M_NOR (0x0 << 11) -#define RT5645_ADC_M_ASYN (0x1 << 11) -#define RT5645_STO_DAC_M_MASK (0x1 << 5) -#define RT5645_STO_DAC_M_SFT 5 -#define RT5645_STO_DAC_M_NOR (0x0 << 5) -#define RT5645_STO_DAC_M_ASYN (0x1 << 5) -#define RT5645_I2S1_R_D_MASK (0x1 << 4) -#define RT5645_I2S1_R_D_SFT 4 -#define RT5645_I2S1_R_D_DIS (0x0 << 4) -#define RT5645_I2S1_R_D_EN (0x1 << 4) -#define RT5645_I2S2_R_D_MASK (0x1 << 3) -#define RT5645_I2S2_R_D_SFT 3 -#define RT5645_I2S2_R_D_DIS (0x0 << 3) -#define RT5645_I2S2_R_D_EN (0x1 << 3) -#define RT5645_PRE_SCLK_MASK (0x3) -#define RT5645_PRE_SCLK_SFT 0 -#define RT5645_PRE_SCLK_512 (0x0) -#define RT5645_PRE_SCLK_1024 (0x1) -#define RT5645_PRE_SCLK_2048 (0x2) +#define RT5645_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5645_DA_STO_CLK_SEL_SFT 12 +#define RT5645_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5645_DA_MONOL_CLK_SEL_SFT 8 +#define RT5645_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5645_DA_MONOR_CLK_SEL_SFT 4 +#define RT5645_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5645_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5645_I2S1_RATE_MASK (0xf << 12) -#define RT5645_I2S1_RATE_SFT 12 -#define RT5645_I2S2_RATE_MASK (0xf << 8) -#define RT5645_I2S2_RATE_SFT 8 +#define RT5645_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5645_AD_MONOL_CLK_SEL_SFT 4 +#define RT5645_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5645_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5645_I2S1_PD_MASK (0x7 << 12) @@ -2175,6 +2161,24 @@ enum { RT5645_DMIC_DATA_GPIO11, }; +enum { + CODEC_TYPE_RT5645, + CODEC_TYPE_RT5650, +}; + +/* filter mask */ +enum { + RT5645_DA_STEREO_FILTER = 0x1, + RT5645_DA_MONO_L_FILTER = (0x1 << 1), + RT5645_DA_MONO_R_FILTER = (0x1 << 2), + RT5645_AD_STEREO_FILTER = (0x1 << 3), + RT5645_AD_MONO_L_FILTER = (0x1 << 4), + RT5645_AD_MONO_R_FILTER = (0x1 << 5), +}; + +int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; @@ -2184,6 +2188,7 @@ struct rt5645_priv { struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; + int codec_type; int sysclk; int sysclk_src; int lrck[RT5645_AIFS]; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index bb0a3ab5416c..9f4c7be6d798 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -376,7 +376,7 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); int idx = -EINVAL; @@ -394,9 +394,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sysclk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int val; - val = snd_soc_read(source->codec, RT5651_GLB_CLK); + val = snd_soc_read(codec, RT5651_GLB_CLK); val &= RT5651_SCLK_SRC_MASK; if (val == RT5651_SCLK_SRC_PLL1) return 1; @@ -731,7 +732,7 @@ static const struct snd_kcontrol_new rt5651_pdm_r_mux = static int rt5651_amp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -769,7 +770,7 @@ static int rt5651_amp_power_event(struct snd_soc_dapm_widget *w, static int rt5651_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -813,7 +814,8 @@ static int rt5651_hp_event(struct snd_soc_dapm_widget *w, static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -833,7 +835,7 @@ static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w, static int rt5651_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -856,7 +858,7 @@ static int rt5651_bst1_event(struct snd_soc_dapm_widget *w, static int rt5651_bst2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -879,7 +881,7 @@ static int rt5651_bst2_event(struct snd_soc_dapm_widget *w, static int rt5651_bst3_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a0833de1665..7b3d6b5992f1 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,10 +14,12 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -498,7 +500,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); int idx = -EINVAL; @@ -515,9 +517,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int val; - val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val = snd_soc_read(codec, RT5670_GLB_CLK); val &= RT5670_SCLK_SRC_MASK; if (val == RT5670_SCLK_SRC_PLL1) return 1; @@ -528,6 +531,7 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg, shift, val; switch (source->shift) { @@ -563,7 +567,7 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, return 0; } - val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + val = (snd_soc_read(codec, reg) >> shift) & 0xf; switch (val) { case 1: case 2: @@ -588,6 +592,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } + +/** + * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0, asrc2_value = 0; + unsigned int asrc3_mask = 0, asrc3_value = 0; + + if (clk_src > RT5670_CLK_SEL_SYS3) + return -EINVAL; + + if (filter_mask & RT5670_DA_STEREO_FILTER) { + asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5670_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_L_FILTER) { + asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_R_FILTER) { + asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_STEREO_FILTER) { + asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_L_FILTER) { + asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_R_FILTER) { + asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_UP_RATE_FILTER) { + asrc3_mask |= RT5670_UP_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK) + | (clk_src << RT5670_UP_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DOWN_RATE_FILTER) { + asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK) + | (clk_src << RT5670_DOWN_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5670_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5670_ASRC_3, + asrc3_mask, asrc3_value); + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, @@ -1146,7 +1233,7 @@ static const struct snd_kcontrol_new rt5670_vad_adc_mux = static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1182,7 +1269,7 @@ static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, static int rt5670_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1232,7 +1319,7 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w, static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1255,7 +1342,7 @@ static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -2188,6 +2275,13 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) return 0; + if (rt5670->pdata.jd_mode) { + if (clk_id == RT5670_SCLK_S_PLL1) + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_sync(&codec->dapm); + } switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2549,6 +2643,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2673,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); @@ -2609,6 +2720,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, RT5670_PWR_MB, RT5670_PWR_MB); regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, @@ -2716,18 +2831,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c207e26..21f8e18c13c4 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) @@ -1983,6 +1966,21 @@ enum { RT5670_DMIC_DATA_GPIO5, }; +/* filter mask */ +enum { + RT5670_DA_STEREO_FILTER = 0x1, + RT5670_DA_MONO_L_FILTER = (0x1 << 1), + RT5670_DA_MONO_R_FILTER = (0x1 << 2), + RT5670_AD_STEREO_FILTER = (0x1 << 3), + RT5670_AD_MONO_L_FILTER = (0x1 << 4), + RT5670_AD_MONO_R_FILTER = (0x1 << 5), + RT5670_UP_RATE_FILTER = (0x1 << 6), + RT5670_DOWN_RATE_FILTER = (0x1 << 7), +}; + +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe1464d268..26fc538f03b1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -784,8 +784,8 @@ static unsigned int bst_tlv[] = { static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; @@ -795,8 +795,9 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; @@ -895,7 +896,7 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); int idx = rl6231_calc_dmic_clk(rt5677->sysclk); @@ -910,7 +911,8 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(source->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int val; regmap_read(rt5677->regmap, RT5677_GLB_CLK1, &val); @@ -921,6 +923,101 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int reg, shift, val; + + if (source->reg == RT5677_ASRC_1) { + switch (source->shift) { + case 12: + reg = RT5677_ASRC_4; + shift = 0; + break; + case 13: + reg = RT5677_ASRC_4; + shift = 4; + break; + case 14: + reg = RT5677_ASRC_4; + shift = 8; + break; + case 15: + reg = RT5677_ASRC_4; + shift = 12; + break; + default: + return 0; + } + } else { + switch (source->shift) { + case 0: + reg = RT5677_ASRC_6; + shift = 8; + break; + case 1: + reg = RT5677_ASRC_6; + shift = 12; + break; + case 2: + reg = RT5677_ASRC_5; + shift = 0; + break; + case 3: + reg = RT5677_ASRC_5; + shift = 4; + break; + case 4: + reg = RT5677_ASRC_5; + shift = 8; + break; + case 5: + reg = RT5677_ASRC_5; + shift = 12; + break; + case 12: + reg = RT5677_ASRC_3; + shift = 0; + break; + case 13: + reg = RT5677_ASRC_3; + shift = 4; + break; + case 14: + reg = RT5677_ASRC_3; + shift = 12; + break; + default: + return 0; + } + } + + regmap_read(rt5677->regmap, reg, &val); + val = (val >> shift) & 0xf; + + switch (val) { + case 1 ... 6: + return 1; + default: + return 0; + } + +} + +static int can_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + if (rt5677->sysclk > rt5677->lrck[RT5677_AIF1] * 384) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, @@ -2030,7 +2127,7 @@ static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux = static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -2054,7 +2151,7 @@ static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, static int rt5677_bst2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -2078,14 +2175,18 @@ static int rt5677_bst2_event(struct snd_soc_dapm_widget *w, static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2096,14 +2197,18 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2114,7 +2219,7 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -2141,7 +2246,7 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int value; @@ -2164,7 +2269,7 @@ static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int value; @@ -2187,7 +2292,7 @@ static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, static int rt5677_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -2211,9 +2316,50 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, - 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT, - 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5677_ASRC_1, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5677_ASRC_1, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S3 ASRC", 1, RT5677_ASRC_1, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S4 ASRC", 1, RT5677_ASRC_1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5677_ASRC_2, 14, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO2 L ASRC", 1, RT5677_ASRC_2, 13, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO2 R ASRC", 1, RT5677_ASRC_2, 12, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO3 L ASRC", 1, RT5677_ASRC_1, 15, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO3 R ASRC", 1, RT5677_ASRC_1, 14, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO4 L ASRC", 1, RT5677_ASRC_1, 13, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO4 R ASRC", 1, RT5677_ASRC_1, 12, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5677_ASRC_2, 11, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO2 ASRC", 1, RT5677_ASRC_2, 10, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO3 ASRC", 1, RT5677_ASRC_2, 9, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO4 ASRC", 1, RT5677_ASRC_2, 8, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5677_ASRC_2, 7, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5677_ASRC_2, 6, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5677_ASRC_2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5677_ASRC_2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO3 ASRC", 1, RT5677_ASRC_2, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO4 ASRC", 1, RT5677_ASRC_2, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5677_ASRC_2, 1, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5677_ASRC_2, 0, 0, NULL, + 0), /* Input Side */ /* micbias */ @@ -2645,10 +2791,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), @@ -2721,6 +2875,31 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc }, + { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc }, + { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", can_use_asrc }, + { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", can_use_asrc }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc }, + { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, + { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, + { "I2S3", NULL, "I2S3 ASRC", can_use_asrc}, + { "I2S4", NULL, "I2S4 ASRC", can_use_asrc}, + + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + { "dac mono2 left filter", NULL, "DAC MONO2 L ASRC", is_using_asrc }, + { "dac mono2 right filter", NULL, "DAC MONO2 R ASRC", is_using_asrc }, + { "dac mono3 left filter", NULL, "DAC MONO3 L ASRC", is_using_asrc }, + { "dac mono3 right filter", NULL, "DAC MONO3 R ASRC", is_using_asrc }, + { "dac mono4 left filter", NULL, "DAC MONO4 L ASRC", is_using_asrc }, + { "dac mono4 right filter", NULL, "DAC MONO4 R ASRC", is_using_asrc }, + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc stereo3 filter", NULL, "ADC STO3 ASRC", is_using_asrc }, + { "adc stereo4 filter", NULL, "ADC STO4 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "DMIC1", NULL, "DMIC L1" }, { "DMIC1", NULL, "DMIC R1" }, { "DMIC2", NULL, "DMIC L2" }, @@ -2851,8 +3030,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, - { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, - { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, @@ -2873,8 +3050,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, { "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" }, - { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, - { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, { "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" }, { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, @@ -2889,8 +3064,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Stereo3 ADC MIXL", NULL, "Sto3 ADC MIXL" }, { "Stereo3 ADC MIXL", NULL, "adc stereo3 filter" }, - { "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll }, - { "Stereo3 ADC MIXR", NULL, "Sto3 ADC MIXR" }, { "Stereo3 ADC MIXR", NULL, "adc stereo3 filter" }, { "adc stereo3 filter", NULL, "PLL1", is_sys_clk_from_pll }, @@ -2905,8 +3078,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Stereo4 ADC MIXL", NULL, "Sto4 ADC MIXL" }, { "Stereo4 ADC MIXL", NULL, "adc stereo4 filter" }, - { "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll }, - { "Stereo4 ADC MIXR", NULL, "Sto4 ADC MIXR" }, { "Stereo4 ADC MIXR", NULL, "adc stereo4 filter" }, { "adc stereo4 filter", NULL, "PLL1", is_sys_clk_from_pll }, @@ -3455,10 +3626,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXL", NULL, "dac stereo1 filter" }, { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXR", NULL, "dac stereo1 filter" }, { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, @@ -3525,35 +3694,46 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Stereo DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Stereo DAC MIXR", "DAC1 L Switch", "DAC1 MIXL" }, { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Mono DAC MIXL", "ST L Switch", "Sidetone Mux" }, { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, - { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXL", NULL, "dac mono2 left filter" }, + { "dac mono2 left filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, - { "Mono DAC MIXR", NULL, "dac mono right filter" }, + { "Mono DAC MIXR", NULL, "dac mono2 right filter" }, + { "dac mono2 right filter", NULL, "PLL1", is_sys_clk_from_pll }, { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXL", NULL, "dac mono3 left filter" }, + { "dac mono3 left filter", NULL, "PLL1", is_sys_clk_from_pll }, { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", NULL, "dac mono3 right filter" }, + { "dac mono3 right filter", NULL, "PLL1", is_sys_clk_from_pll }, { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXL", NULL, "dac mono4 left filter" }, + { "dac mono4 left filter", NULL, "PLL1", is_sys_clk_from_pll }, { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", NULL, "dac mono4 right filter" }, + { "dac mono4 right filter", NULL, "PLL1", is_sys_clk_from_pll }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, @@ -3575,11 +3755,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC3 SRC Mux", "DD MIX2L", "DD2 MIXL" }, { "DAC 1", NULL, "DAC12 SRC Mux" }, - { "DAC 1", NULL, "PLL1", is_sys_clk_from_pll }, { "DAC 2", NULL, "DAC12 SRC Mux" }, - { "DAC 2", NULL, "PLL1", is_sys_clk_from_pll }, { "DAC 3", NULL, "DAC3 SRC Mux" }, - { "DAC 3", NULL, "PLL1", is_sys_clk_from_pll }, { "PDM1 L Mux", "STO1 DAC MIX", "Stereo DAC MIXL" }, { "PDM1 L Mux", "MONO DAC MIX", "Mono DAC MIXL" }, @@ -3926,7 +4103,8 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val = 0; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, slot_width_25 = 0; if (rx_mask || tx_mask) val |= (1 << 12); @@ -3950,6 +4128,8 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, case 20: val |= (1 << 8); break; + case 25: + slot_width_25 = 0x8080; case 24: val |= (2 << 8); break; @@ -3963,10 +4143,16 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (dai->id) { case RT5677_AIF1: - snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, 0x1f00, + val); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x8000, + slot_width_25); break; case RT5677_AIF2: - snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, 0x1f00, + val); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x80, + slot_width_25); break; default: break; @@ -4751,6 +4937,11 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, RT5677_GPIO5_DIR_OUT); } + if (rt5677->pdata.micbias1_vdd_3v3) + regmap_update_bits(rt5677->regmap, RT5677_MICBIAS, + RT5677_MICBIAS1_CTRL_VDD_MASK, + RT5677_MICBIAS1_CTRL_VDD_3_3V); + rt5677_init_gpio(i2c); rt5677_init_irq(i2c); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce610f4..e182e6569bbd 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -155,18 +155,19 @@ struct sgtl5000_priv { static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias resistor */ - snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, 0); break; } @@ -181,11 +182,12 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; switch (event) { case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; @@ -195,9 +197,9 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, * operational to prevent inadvertently starving the * other one of them. */ - if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) & mask) != mask) { - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); } @@ -483,21 +485,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* setting i2s data format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_DSP_B: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; case SND_SOC_DAIFMT_I2S: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_RIGHT_J: - i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRPOL; break; case SND_SOC_DAIFMT_LEFT_J: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; default: @@ -1462,6 +1464,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + udelay(1); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); if (ret) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 1f451a1946eb..47b257e41809 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -233,16 +233,18 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); /* power up the rail */ - snd_soc_write(w->codec, SN95031_VHSP, 0x3D); - snd_soc_write(w->codec, SN95031_VHSN, 0x3F); + snd_soc_write(codec, SN95031_VHSP, 0x3D); + snd_soc_write(codec, SN95031_VHSN, 0x3F); msleep(1); } else if (SND_SOC_DAPM_EVENT_OFF(event)) { pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); - snd_soc_write(w->codec, SN95031_VHSP, 0xC4); - snd_soc_write(w->codec, SN95031_VHSN, 0x04); + snd_soc_write(codec, SN95031_VHSP, 0xC4); + snd_soc_write(codec, SN95031_VHSN, 0x04); } return 0; } @@ -250,14 +252,16 @@ static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); /* power up the rail */ - snd_soc_write(w->codec, SN95031_VIHF, 0x27); + snd_soc_write(codec, SN95031_VIHF, 0x27); msleep(1); } else if (SND_SOC_DAPM_EVENT_OFF(event)) { pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); - snd_soc_write(w->codec, SN95031_VIHF, 0x24); + snd_soc_write(codec, SN95031_VIHF, 0x24); } return 0; } @@ -265,6 +269,7 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int ldo = 0, clk_dir = 0, data_dir = 0; if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -273,15 +278,16 @@ static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, data_dir = BIT(7); } /* program DMIC LDO, clock and set clock */ - snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); - snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir); - snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(0), clk_dir); + snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(7), data_dir); return 0; } static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int ldo = 0, clk_dir = 0, data_dir = 0; if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -290,22 +296,23 @@ static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, data_dir = BIT(1); } /* program DMIC LDO, clock and set clock */ - snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); - snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir); - snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(2), clk_dir); + snd_soc_update_bits(codec, SN95031_DMICBUF45, BIT(1), data_dir); return 0; } static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int ldo = 0; if (SND_SOC_DAPM_EVENT_ON(event)) ldo = BIT(7)|BIT(6); /* program DMIC LDO */ - snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); return 0; } diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index d8e32a6262ee..d3191c983d71 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -131,7 +131,7 @@ #define STA32X_CONFF_OCFG_MASK 0x03 #define STA32X_CONFF_OCFG_SHIFT 0 #define STA32X_CONFF_IDE 0x04 -#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_IDE_SHIFT 2 #define STA32X_CONFF_BCLE 0x08 #define STA32X_CONFF_ECLE 0x20 #define STA32X_CONFF_PWDN 0x40 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index dc3223d6eca1..c86dd9aae157 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -349,7 +349,8 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); unsigned int reg = AIC31XX_DACFLAG1; unsigned int mask; @@ -377,7 +378,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, reg = AIC31XX_ADCFLAG; break; default: - dev_err(w->codec->dev, "Unknown widget '%s' calling %s\n", + dev_err(codec->dev, "Unknown widget '%s' calling %s\n", w->name, __func__); return -EINVAL; } @@ -388,7 +389,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); default: - dev_dbg(w->codec->dev, + dev_dbg(codec->dev, "Unhandled dapm widget event %d from %s\n", event, w->name); } @@ -433,7 +434,7 @@ static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); switch (event) { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce054b4e..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -197,7 +197,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 0fe2ced5b09f..4e3e607dec13 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -423,17 +423,18 @@ exit: static int dac33_playback_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream, w->codec); - dac33_prepare_chip(dac33->substream, w->codec); + dac33_calculate_times(dac33->substream, codec); + dac33_prepare_chip(dac33->substream, codec); } break; case SND_SOC_DAPM_POST_PMD: - dac33_disable_digital(w->codec); + dac33_disable_digital(codec); break; } return 0; diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 1d1205702d23..9f2dced046de 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, struct ts3a227e *ts3a227e; struct device *dev = &i2c->dev; int ret; + unsigned int acc_reg; ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); if (ts3a227e == NULL) @@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, ADC_COMPLETE_INT_DISABLE); + /* Read jack status because chip might not trigger interrupt at boot. */ + regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + ts3a227e_jack_report(ts3a227e); + return 0; } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 44af3188afb9..d04693e9cf9f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -567,12 +567,13 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); \ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ twl4030->pin_name##_enabled = 1; \ - twl4030_write(w->codec, reg, twl4030_read(w->codec, reg)); \ + twl4030_write(codec, reg, twl4030_read(codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ twl4030->pin_name##_enabled = 0; \ @@ -621,12 +622,14 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) static int handsfreelpga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_POST_PMU: - handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1); + handsfree_ramp(codec, TWL4030_REG_HFL_CTL, 1); break; case SND_SOC_DAPM_POST_PMD: - handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0); + handsfree_ramp(codec, TWL4030_REG_HFL_CTL, 0); break; } return 0; @@ -635,12 +638,14 @@ static int handsfreelpga_event(struct snd_soc_dapm_widget *w, static int handsfreerpga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_POST_PMU: - handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1); + handsfree_ramp(codec, TWL4030_REG_HFR_CTL, 1); break; case SND_SOC_DAPM_POST_PMD: - handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0); + handsfree_ramp(codec, TWL4030_REG_HFR_CTL, 0); break; } return 0; @@ -649,19 +654,23 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static int vibramux_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + twl4030_write(codec, TWL4030_REG_VIBRA_SET, 0xff); return 0; } static int apll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_PRE_PMU: - twl4030_apll_enable(w->codec, 1); + twl4030_apll_enable(codec, 1); break; case SND_SOC_DAPM_POST_PMD: - twl4030_apll_enable(w->codec, 0); + twl4030_apll_enable(codec, 0); break; } return 0; @@ -670,23 +679,24 @@ static int apll_event(struct snd_soc_dapm_widget *w, static int aif_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u8 audio_if; - audio_if = twl4030_read(w->codec, TWL4030_REG_AUDIO_IF); + audio_if = twl4030_read(codec, TWL4030_REG_AUDIO_IF); switch (event) { case SND_SOC_DAPM_PRE_PMU: /* Enable AIF */ /* enable the PLL before we use it to clock the DAI */ - twl4030_apll_enable(w->codec, 1); + twl4030_apll_enable(codec, 1); - twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, + twl4030_write(codec, TWL4030_REG_AUDIO_IF, audio_if | TWL4030_AIF_EN); break; case SND_SOC_DAPM_POST_PMD: /* disable the DAI before we stop it's source PLL */ - twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, + twl4030_write(codec, TWL4030_REG_AUDIO_IF, audio_if & ~TWL4030_AIF_EN); - twl4030_apll_enable(w->codec, 0); + twl4030_apll_enable(codec, 0); break; } return 0; @@ -758,20 +768,21 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) static int headsetlpga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: /* Do the ramp-up only once */ if (!twl4030->hsr_enabled) - headset_ramp(w->codec, 1); + headset_ramp(codec, 1); twl4030->hsl_enabled = 1; break; case SND_SOC_DAPM_POST_PMD: /* Do the ramp-down only if both headsetL/R is disabled */ if (!twl4030->hsr_enabled) - headset_ramp(w->codec, 0); + headset_ramp(codec, 0); twl4030->hsl_enabled = 0; break; @@ -782,20 +793,21 @@ static int headsetlpga_event(struct snd_soc_dapm_widget *w, static int headsetrpga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: /* Do the ramp-up only once */ if (!twl4030->hsl_enabled) - headset_ramp(w->codec, 1); + headset_ramp(codec, 1); twl4030->hsr_enabled = 1; break; case SND_SOC_DAPM_POST_PMD: /* Do the ramp-down only if both headsetL/R is disabled */ if (!twl4030->hsl_enabled) - headset_ramp(w->codec, 0); + headset_ramp(codec, 0); twl4030->hsr_enabled = 0; break; @@ -806,7 +818,8 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, static int digimic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct twl4030_codec_data *pdata = twl4030->pdata; if (pdata && pdata->digimic_delay) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 90f47f988b3f..aeec27b6f1af 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -234,7 +234,7 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u8 hslctl, hsrctl; /* @@ -261,7 +261,7 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, static int twl6040_ep_drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret = 0; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 34ef65c52a7d..8d9de49a5052 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -683,7 +683,7 @@ static const struct snd_kcontrol_new wm2000_controls[] = { static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b80970dc2d2f..ea09db585aa1 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -775,7 +775,8 @@ static int wm5100_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); switch (w->reg) { case WM5100_CHANNEL_ENABLES_1: @@ -839,7 +840,7 @@ static int wm5100_post_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f439ae052128..6d0fe0ac95a3 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -580,7 +581,7 @@ static const struct reg_default wm5102_sysclk_revb_patch[] = { static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; @@ -617,11 +618,10 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&arizona->dac_comp_lock); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&arizona->dac_comp_lock); return 0; @@ -1272,19 +1272,24 @@ SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4456b38a3ef5..fbaeddb3e903 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -134,7 +134,7 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = { static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; @@ -905,22 +905,28 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT3R", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 574579b98872..c81a9eab3e3e 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -259,7 +259,7 @@ static void wm8350_pga_work(struct work_struct *work) static int pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 8ee446987aa9..b0d84e552fca 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -324,6 +324,7 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; u32 reg_shift = mc->shift; @@ -332,7 +333,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : - reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1); + reg = snd_soc_read(codec, WM8400_OUTPUT_MIXER1); if (reg & WM8400_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -340,7 +341,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): - reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2); + reg = snd_soc_read(codec, WM8400_OUTPUT_MIXER2); if (reg & WM8400_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -348,7 +349,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): - reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -356,7 +357,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): - reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b42f6e9..098c143f44d6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -217,7 +217,8 @@ SND_SOC_DAPM_INPUT("LLINEIN"), static int wm8731_check_osc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); return wm8731->sysclk_type == WM8731_SYSCLK_XTAL; } @@ -717,6 +718,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f6847fdd6ddd..eb0a1644ba11 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -323,7 +323,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT2"), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), - SND_SOC_DAPM_OUTPUT("VREF"), + SND_SOC_DAPM_VMID("VREF"), SND_SOC_DAPM_INPUT("LINPUT1"), SND_SOC_DAPM_INPUT("LINPUT2"), diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 180e7a098726..53e977da2f86 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -308,9 +308,7 @@ static const struct snd_soc_dapm_route wm8770_intercon[] = { static int vout12supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec; - - codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -327,9 +325,7 @@ static int vout12supply_event(struct snd_soc_dapm_widget *w, static int vout34supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec; - - codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3a0d4b7d692f..2eb986c19b88 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -224,7 +224,7 @@ static void wm8900_reset(struct snd_soc_codec *codec) static int wm8900_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 hpctl1 = snd_soc_read(codec, WM8900_REG_HPCTL1); switch (event) { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index cc6b0ef98a34..dde462c082be 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -260,7 +260,7 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); switch (event) { diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1380d5..c5eaa0198ef0 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -673,7 +673,7 @@ static int cp_event(struct snd_soc_dapm_widget *w, static int sysclk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -711,7 +711,7 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, static int out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); int reg, val; int dcs_mask; @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL Mux", "Left", "AIFINL" }, + { "DACL Mux", "Right", "AIFINR" }, + + { "DACR Mux", "Left", "AIFINL" }, + { "DACR Mux", "Right", "AIFINR" }, + + { "DACL", NULL, "DACL Mux" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "DACR Mux" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 1173f7fef5a7..1ab2d462afad 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -333,7 +333,7 @@ static int wm8955_configure_clocking(struct snd_soc_codec *codec) static int wm8955_sysclk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int ret = 0; /* Always disable the clocks - if we're doing reconfiguration this diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 3cbc82b33292..c799cca5abeb 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -418,7 +418,7 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int i; switch (event) { diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae71d94..a96eb497a379 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -556,7 +556,7 @@ static struct { { 22050, 2 }, { 24000, 2 }, { 16000, 3 }, - { 11250, 4 }, + { 11025, 4 }, { 12000, 4 }, { 8000, 5 }, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index eeffd05384b4..95e2c1bfc809 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -194,7 +194,7 @@ static bool wm8961_readable(struct device *dev, unsigned int reg) static int wm8961_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 hp_reg = snd_soc_read(codec, WM8961_ANALOGUE_HP_0); u16 cp_reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_1); u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2); @@ -286,7 +286,7 @@ static int wm8961_hp_event(struct snd_soc_dapm_widget *w, static int wm8961_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2); u16 spk_reg = snd_soc_read(codec, WM8961_CLASS_D_CONTROL_1); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d32d554f5b34..118b0034ba23 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1866,7 +1866,7 @@ static int cp_event(struct snd_soc_dapm_widget *w, static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int timeout; int reg; int expected = (WM8962_DCS_STARTUP_DONE_HP1L | @@ -1960,7 +1960,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, static int out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int reg; switch (w->shift) { @@ -1993,7 +1993,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, static int dsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); switch (event) { diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index e418199155a8..baff2cc222a6 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -244,7 +244,7 @@ SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V, static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2); /* Use the DAC to gate LRC if active, otherwise use ADC */ diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 8a584229310a..c93bffcb3cfb 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -374,13 +374,14 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u32 reg_shift = kcontrol->private_value & 0xfff; int ret = 0; u16 reg; switch (reg_shift) { case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) : - reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER1); + reg = snd_soc_read(codec, WM8990_OUTPUT_MIXER1); if (reg & WM8990_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -388,7 +389,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8): - reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER2); + reg = snd_soc_read(codec, WM8990_OUTPUT_MIXER2); if (reg & WM8990_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -396,7 +397,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8): - reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8990_SPEAKER_MIXER); if (reg & WM8990_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -404,7 +405,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8): - reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8990_SPEAKER_MIXER); if (reg & WM8990_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index b0ac2c3e31b9..49df0dc607e6 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -382,13 +382,14 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = { static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u32 reg_shift = kcontrol->private_value & 0xfff; int ret = 0; u16 reg; switch (reg_shift) { case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8): - reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1); + reg = snd_soc_read(codec, WM8991_OUTPUT_MIXER1); if (reg & WM8991_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -397,7 +398,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, break; case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8): - reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2); + reg = snd_soc_read(codec, WM8991_OUTPUT_MIXER2); if (reg & WM8991_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -406,7 +407,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, break; case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8): - reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8991_SPEAKER_MIXER); if (reg & WM8991_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -415,7 +416,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, break; case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8): - reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + reg = snd_soc_read(codec, WM8991_SPEAKER_MIXER); if (reg & WM8991_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 53c6fe359496..2e70a270eb28 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -810,7 +810,7 @@ SOC_SINGLE_TLV("EQ5 Volume", WM8993_EQ6, 0, 24, 0, eq_tlv), static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1b97de2e4e67..4fbc7689339a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -249,7 +249,8 @@ static int configure_clock(struct snd_soc_codec *codec) static int check_clk_sys(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - int reg = snd_soc_read(source->codec, WM8994_CLOCKING_1); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + int reg = snd_soc_read(codec, WM8994_CLOCKING_1); const char *clk; /* Check what we're currently using for CLK_SYS */ @@ -806,7 +807,7 @@ static void active_dereference(struct snd_soc_codec *codec) static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -981,7 +982,7 @@ static void vmid_dereference(struct snd_soc_codec *codec) static int vmid_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1037,7 +1038,7 @@ static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec) static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; @@ -1135,7 +1136,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int i; int dac; int adc; @@ -1220,7 +1221,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1238,7 +1239,7 @@ static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1256,7 +1257,7 @@ static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, static int late_enable_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1289,7 +1290,7 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, static int late_disable_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -1331,7 +1332,7 @@ static int micbias_ev(struct snd_soc_dapm_widget *w, static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int mask = 1 << w->shift; snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, @@ -1372,7 +1373,7 @@ SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 0, 1, 0), static int post_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); dev_dbg(codec->dev, "SRC status: %x\n", snd_soc_read(codec, WM8994_RATE_STATUS)); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index c280f0a3a424..79e1aead5131 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -534,10 +534,11 @@ static void wm8995_update_class_w(struct snd_soc_codec *codec) static int check_clk_sys(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); unsigned int reg; const char *clk; - reg = snd_soc_read(source->codec, WM8995_CLOCKING_1); + reg = snd_soc_read(codec, WM8995_CLOCKING_1); /* Check what we're currently using for CLK_SYS */ if (reg & WM8995_SYSCLK_SRC) clk = "AIF2CLK"; @@ -560,9 +561,7 @@ static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec; - - codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -611,10 +610,9 @@ static void dc_servo_cmd(struct snd_soc_codec *codec, static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int reg; - codec = w->codec; reg = snd_soc_read(codec, WM8995_ANALOGUE_HP_1); switch (event) { @@ -761,9 +759,7 @@ static int configure_clock(struct snd_soc_codec *codec) static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec; - - codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_PRE_PMU: diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index b1dcc11c1b23..dc92d5e4e942 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -599,7 +599,7 @@ static void wm8996_bg_disable(struct snd_soc_codec *codec) static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int ret = 0; switch (event) { @@ -634,7 +634,8 @@ static int cp_event(struct snd_soc_dapm_widget *w, static int rmv_short_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); /* Record which outputs we enabled */ switch (event) { @@ -758,7 +759,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, static int dcs_start(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 7e8bfe27566b..a4d11770630c 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -84,7 +84,7 @@ static const struct reg_default wm8997_sysclk_reva_patch[] = { static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; @@ -610,13 +610,16 @@ SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b1d946facd57..13a3f335ea5b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -734,7 +734,7 @@ static int configure_clock(struct snd_soc_codec *codec) static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); /* This should be done on init() for bypass paths */ diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 6ffe8dc4f3fa..60d243c904f5 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -254,7 +254,7 @@ SOC_SINGLE_TLV("MIXOUTR IN2B Volume", WM9090_OUTPUT_MIXER4, 0, 3, 1, static int hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int reg = snd_soc_read(codec, WM9090_ANALOGUE_HP_0); switch (event) { diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3eddb18fefd1..5cc457ef8894 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -344,23 +344,27 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) struct snd_ac97 *ac97; int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } - snd_soc_codec_set_drvdata(codec, ac97); - ret = wm9705_reset(codec); if (ret) - goto reset_err; + goto err_put_device; + + ret = device_add(&ac97->dev); + if (ret) + goto err_put_device; + + snd_soc_codec_set_drvdata(codec, ac97); return 0; -reset_err: - snd_soc_free_ac97_codec(ac97); +err_put_device: + put_device(&ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e04643d2bb24..9517571e820d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -666,7 +666,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int ret = 0; - wm9712->ac97 = snd_soc_new_ac97_codec(codec); + wm9712->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); @@ -675,15 +675,19 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ret = wm9712_reset(codec, 0); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9712->ac97->dev); + if (ret) + goto err_put_device; /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; -reset_err: - snd_soc_free_ac97_codec(wm9712->ac97); +err_put_device: + put_device(&wm9712->ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 71b9d5b0734d..68222917b396 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -217,7 +217,7 @@ SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 status, rate; if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD)) @@ -1225,7 +1225,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - wm9713->ac97 = snd_soc_new_ac97_codec(codec); + wm9713->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); @@ -1234,7 +1234,11 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9713->ac97->dev); + if (ret) + goto err_put_device; /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; @@ -1242,8 +1246,8 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return 0; -reset_err: - snd_soc_free_ac97_codec(wm9713->ac97); +err_put_device: + put_device(&wm9713->ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 720d6e852986..ff67b334065b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1373,7 +1373,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; @@ -1605,7 +1605,7 @@ err: int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; @@ -1626,7 +1626,7 @@ EXPORT_SYMBOL_GPL(wm_adsp2_early_event); int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 374537d5e179..8366e19657a7 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -500,7 +500,7 @@ SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1, static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -542,7 +542,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int reg = snd_soc_read(codec, WM8993_ANALOGUE_HP_0); switch (event) { @@ -594,7 +594,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, static int earpiece_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); u16 reg = snd_soc_read(codec, WM8993_ANTIPOP1) & ~WM8993_HPOUT2_IN_ENA; switch (event) { @@ -619,7 +619,7 @@ static int earpiece_event(struct snd_soc_dapm_widget *w, static int lineout_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); bool *flag; @@ -649,7 +649,7 @@ static int lineout_event(struct snd_soc_dapm_widget *w, static int micbias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); switch (w->shift) { diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8e948c63f3d9..2b81ca418d2a 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -58,13 +58,12 @@ choice depends on MACH_DAVINCI_DM365_EVM config SND_DM365_AIC3X_CODEC - bool "Audio Codec - AIC3101" + tristate "Audio Codec - AIC3101" help Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC tristate "Voice Codec - CQ93VC" - depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 158cb3d1db70..b6bb5947a8a8 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -14,7 +14,6 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/platform_data/edma.h> #include <linux/i2c.h> #include <linux/of_platform.h> #include <linux/clk.h> @@ -25,11 +24,6 @@ #include <asm/dma.h> #include <asm/mach-types.h> -#include <linux/edma.h> - -#include "davinci-pcm.h" -#include "davinci-i2s.h" - struct snd_soc_card_drvdata_davinci { struct clk *mclk; unsigned sysclk; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4f9c5d..de3b155a5011 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) return IRQ_RETVAL(handled_mask); } +static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + irqreturn_t ret = IRQ_NONE; + + if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]) + ret = davinci_mcasp_tx_irq_handler(irq, data); + + if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]) + ret |= davinci_mcasp_rx_irq_handler(irq, data); + + return ret; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -1313,16 +1327,19 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( pdata->tx_dma_channel = dma_spec.args[0]; - ret = of_property_match_string(np, "dma-names", "rx"); - if (ret < 0) - goto nodata; + /* RX is not valid in DIT mode */ + if (pdata->op_mode != DAVINCI_MCASP_DIT_MODE) { + ret = of_property_match_string(np, "dma-names", "rx"); + if (ret < 0) + goto nodata; - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, - &dma_spec); - if (ret < 0) - goto nodata; + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; - pdata->rx_dma_channel = dma_spec.args[0]; + pdata->rx_dma_channel = dma_spec.args[0]; + } ret = of_property_read_u32(np, "tx-num-evt", &val); if (ret >= 0) @@ -1441,6 +1458,23 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "common"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_common_irq_handler, + IRQF_ONESHOT | IRQF_SHARED, + irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "common IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", @@ -1501,19 +1535,34 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->filter_data = &dma_params->channel; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; - if (dat) - dma_params->dma_addr = dat->start; - else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + /* RX is not valid in DIT mode */ + if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_capture; + if (dat) + dma_params->dma_addr = dat->start; + else + dma_params->dma_addr = mem->start + pdata->rx_dma_offset; + + /* Unconditional dmaengine stuff */ + dma_data->addr = dma_params->dma_addr; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (res) + dma_params->channel = res->start; + else + dma_params->channel = pdata->rx_dma_channel; + + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "rx"; + else + dma_data->filter_data = &dma_params->channel; + } if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; @@ -1523,18 +1572,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res) - dma_params->channel = res->start; - else - dma_params->channel = pdata->rx_dma_channel; - - /* dmaengine filter data for DT and non-DT boot */ - if (pdev->dev.of_node) - dma_data->filter_data = "rx"; - else - dma_data->filter_data = &dma_params->channel; - dev_set_drvdata(&pdev->dev, mcasp); mcasp_reparent_fck(pdev); diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index e334900cf0b8..d50e08517dce 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,6 +1,7 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" depends on CLKDEV_LOOKUP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index b93168d4f648..a3e97b46b64e 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> /* common register for all channel */ #define IER 0x000 @@ -54,9 +55,39 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +/* + * Component parameter register fields - define the I2S block's + * configuration. + */ +#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25) +#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22) +#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19) +#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16) +#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9) +#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7) +#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6) +#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5) +#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4) +#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2) +#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0) + +#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10) +#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7) +#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3) +#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0) + +/* Number of entries in WORDSIZE and DATA_WIDTH parameter registers */ +#define COMP_MAX_WORDSIZE (1 << 3) +#define COMP_MAX_DATA_WIDTH (1 << 2) + #define MAX_CHANNEL_NUM 8 #define MIN_CHANNEL_NUM 2 +union dw_i2s_snd_dma_data { + struct i2s_dma_data pd; + struct snd_dmaengine_dai_dma_data dt; +}; + struct dw_i2s_dev { void __iomem *i2s_base; struct clk *clk; @@ -65,8 +96,8 @@ struct dw_i2s_dev { struct device *dev; /* data related to DMA transfers b/w i2s and DMAC */ - struct i2s_dma_data play_dma_data; - struct i2s_dma_data capture_dma_data; + union dw_i2s_snd_dma_data play_dma_data; + union dw_i2s_snd_dma_data capture_dma_data; struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; @@ -153,7 +184,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - struct i2s_dma_data *dma_data = NULL; + union dw_i2s_snd_dma_data *dma_data = NULL; if (!(dev->capability & DWC_I2S_RECORD) && (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) @@ -209,16 +240,9 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: - ch_reg = 3; - break; case SIX_CHANNEL_SUPPORT: - ch_reg = 2; - break; case FOUR_CHANNEL_SUPPORT: - ch_reg = 1; - break; case TWO_CHANNEL_SUPPORT: - ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); @@ -227,31 +251,43 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + for (ch_reg = 0; ch_reg < (config->chan_nr / 2); ch_reg++) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } } i2s_write_reg(dev->i2s_base, CCR, ccr); config->sample_rate = params_rate(params); - if (!dev->i2s_clk_cfg) - return -EINVAL; + if (dev->i2s_clk_cfg) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk = config->sample_rate * config->data_width * 2; - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; + ret = clk_set_rate(dev->clk, bitclk); + if (ret) { + dev_err(dev->dev, "Can't set I2S clock rate: %d\n", + ret); + return ret; + } } return 0; @@ -263,6 +299,19 @@ static void dw_i2s_shutdown(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, NULL); } +static int dw_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, TXFFR, 1); + else + i2s_write_reg(dev->i2s_base, RXFFR, 1); + + return 0; +} + static int dw_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -294,6 +343,7 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, + .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, }; @@ -324,20 +374,162 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) #define dw_i2s_resume NULL #endif +/* + * The following tables allow a direct lookup of various parameters + * defined in the I2S block's configuration in terms of sound system + * parameters. Each table is sized to the number of entries possible + * according to the number of configuration bits describing an I2S + * block parameter. + */ + +/* Maximum bit resolution of a channel - not uniformly spaced */ +static const u32 fifo_width[COMP_MAX_WORDSIZE] = { + 12, 16, 20, 24, 32, 0, 0, 0 +}; + +/* Width of (DMA) bus */ +static const u32 bus_widths[COMP_MAX_DATA_WIDTH] = { + DMA_SLAVE_BUSWIDTH_1_BYTE, + DMA_SLAVE_BUSWIDTH_2_BYTES, + DMA_SLAVE_BUSWIDTH_4_BYTES, + DMA_SLAVE_BUSWIDTH_UNDEFINED +}; + +/* PCM format to support channel resolution */ +static const u32 formats[COMP_MAX_WORDSIZE] = { + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S32_LE, + 0, + 0, + 0 +}; + +static int dw_configure_dai(struct dw_i2s_dev *dev, + struct snd_soc_dai_driver *dw_i2s_dai, + unsigned int rates) +{ + /* + * Read component parameter registers to extract + * the I2S block's configuration. + */ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 idx; + + if (COMP1_TX_ENABLED(comp1)) { + dev_dbg(dev->dev, " designware: play supported\n"); + idx = COMP1_TX_WORDSIZE_0(comp1); + if (WARN_ON(idx >= ARRAY_SIZE(formats))) + return -EINVAL; + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = + 1 << (COMP1_TX_CHANNELS(comp1) + 1); + dw_i2s_dai->playback.formats = formats[idx]; + dw_i2s_dai->playback.rates = rates; + } + + if (COMP1_RX_ENABLED(comp1)) { + dev_dbg(dev->dev, "designware: record supported\n"); + idx = COMP2_RX_WORDSIZE_0(comp2); + if (WARN_ON(idx >= ARRAY_SIZE(formats))) + return -EINVAL; + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = + 1 << (COMP1_RX_CHANNELS(comp1) + 1); + dw_i2s_dai->capture.formats = formats[idx]; + dw_i2s_dai->capture.rates = rates; + } + + return 0; +} + +static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, + struct snd_soc_dai_driver *dw_i2s_dai, + struct resource *res, + const struct i2s_platform_data *pdata) +{ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 idx = COMP1_APB_DATA_WIDTH(comp1); + int ret; + + if (WARN_ON(idx >= ARRAY_SIZE(bus_widths))) + return -EINVAL; + + ret = dw_configure_dai(dev, dw_i2s_dai, pdata->snd_rates); + if (ret < 0) + return ret; + + /* Set DMA slaves info */ + dev->play_dma_data.pd.data = pdata->play_dma_data; + dev->capture_dma_data.pd.data = pdata->capture_dma_data; + dev->play_dma_data.pd.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA; + dev->play_dma_data.pd.max_burst = 16; + dev->capture_dma_data.pd.max_burst = 16; + dev->play_dma_data.pd.addr_width = bus_widths[idx]; + dev->capture_dma_data.pd.addr_width = bus_widths[idx]; + dev->play_dma_data.pd.filter = pdata->filter; + dev->capture_dma_data.pd.filter = pdata->filter; + + return 0; +} + +static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, + struct snd_soc_dai_driver *dw_i2s_dai, + struct resource *res) +{ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); + u32 idx = COMP1_APB_DATA_WIDTH(comp1); + u32 idx2; + int ret; + + if (WARN_ON(idx >= ARRAY_SIZE(bus_widths))) + return -EINVAL; + + ret = dw_configure_dai(dev, dw_i2s_dai, SNDRV_PCM_RATE_8000_192000); + if (ret < 0) + return ret; + + if (COMP1_TX_ENABLED(comp1)) { + idx2 = COMP1_TX_WORDSIZE_0(comp1); + + dev->capability |= DWC_I2S_PLAY; + dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; + dev->play_dma_data.dt.addr_width = bus_widths[idx]; + dev->play_dma_data.dt.chan_name = "TX"; + dev->play_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[idx2]) >> 8; + dev->play_dma_data.dt.maxburst = 16; + } + if (COMP1_RX_ENABLED(comp1)) { + idx2 = COMP2_RX_WORDSIZE_0(comp2); + + dev->capability |= DWC_I2S_RECORD; + dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; + dev->capture_dma_data.dt.addr_width = bus_widths[idx]; + dev->capture_dma_data.dt.chan_name = "RX"; + dev->capture_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[idx2] >> 8); + dev->capture_dma_data.dt.maxburst = 16; + } + + return 0; + +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; struct dw_i2s_dev *dev; struct resource *res; int ret; - unsigned int cap; struct snd_soc_dai_driver *dw_i2s_dai; - if (!pdata) { - dev_err(&pdev->dev, "Invalid platform data\n"); - return -EINVAL; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); @@ -345,83 +537,67 @@ static int dw_i2s_probe(struct platform_device *pdev) } dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); + if (!dw_i2s_dai) return -ENOMEM; - } dw_i2s_dai->ops = &dw_i2s_dai_ops; dw_i2s_dai->suspend = dw_i2s_suspend; dw_i2s_dai->resume = dw_i2s_resume; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dev->i2s_base)) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + if (IS_ERR(dev->i2s_base)) return PTR_ERR(dev->i2s_base); - } - cap = pdata->cap; - dev->capability = cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + dev->dev = &pdev->dev; + if (pdata) { + ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); + if (ret < 0) + return ret; + + dev->capability = pdata->cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock configure method\n"); + return -ENODEV; + } - /* Set DMA slaves info */ + dev->clk = devm_clk_get(&pdev->dev, NULL); + } else { + ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); + if (ret < 0) + return ret; - dev->play_dma_data.data = pdata->play_dma_data; - dev->capture_dma_data.data = pdata->capture_dma_data; - dev->play_dma_data.addr = res->start + I2S_TXDMA; - dev->capture_dma_data.addr = res->start + I2S_RXDMA; - dev->play_dma_data.max_burst = 16; - dev->capture_dma_data.max_burst = 16; - dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->play_dma_data.filter = pdata->filter; - dev->capture_dma_data.filter = pdata->filter; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); + } if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); + return PTR_ERR(dev->clk); - ret = clk_enable(dev->clk); + ret = clk_prepare_enable(dev->clk); if (ret < 0) - goto err_clk_put; - - if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " designware: play supported\n"); - dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->playback.channels_max = pdata->channel; - dw_i2s_dai->playback.formats = pdata->snd_fmts; - dw_i2s_dai->playback.rates = pdata->snd_rates; - } - - if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "designware: record supported\n"); - dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->capture.channels_max = pdata->channel; - dw_i2s_dai->capture.formats = pdata->snd_fmts; - dw_i2s_dai->capture.rates = pdata->snd_rates; - } + return ret; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); - ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, + ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component, dw_i2s_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "not able to register dai\n"); goto err_clk_disable; } + if (!pdata) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, + "Could not register PCM: %d\n", ret); + goto err_clk_disable; + } + } + return 0; err_clk_disable: - clk_disable(dev->clk); -err_clk_put: - clk_put(dev->clk); + clk_disable_unprepare(dev->clk); return ret; } @@ -429,18 +605,26 @@ static int dw_i2s_remove(struct platform_device *pdev) { struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - clk_put(dev->clk); + clk_disable_unprepare(dev->clk); return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dw_i2s_of_match[] = { + { .compatible = "snps,designware-i2s", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, dw_i2s_of_match); +#endif + static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", + .of_match_table = of_match_ptr(dw_i2s_of_match), }, }; diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9ce70fc67b09..8c9e9006dd84 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,25 +42,6 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - /* fsl_ssi lacks the set_fmt ops. */ - if (ret && ret != -ENOTSUPP) { - dev_err(cpu_dai->dev, - "Failed to set the cpu dai format.\n"); - return ret; - } - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - dev_err(cpu_dai->dev, - "Failed to set the codec format.\n"); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { @@ -91,6 +72,8 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &eukrea_tlv320_snd_ops, }; diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 026a80117540..c068494bae30 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -818,7 +818,6 @@ static int fsl_asrc_probe(struct platform_device *pdev) return -ENOMEM; asrc_priv->pdev = pdev; - strncpy(asrc_priv->name, np->name, sizeof(asrc_priv->name) - 1); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -837,12 +836,12 @@ static int fsl_asrc_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return irq; } ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0, - asrc_priv->name, asrc_priv); + dev_name(&pdev->dev), asrc_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret); return ret; diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index a3f211f53c23..4aed63c4b431 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -433,7 +433,6 @@ struct fsl_asrc_pair { * @channel_avail: non-occupied channel numbers * @asrc_rate: default sample rate for ASoC Back-Ends * @asrc_width: default sample width for ASoC Back-Ends - * @name: driver name */ struct fsl_asrc { struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -452,8 +451,6 @@ struct fsl_asrc { int asrc_rate; int asrc_width; - - char name[32]; }; extern struct snd_soc_platform_driver fsl_asrc_platform; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 1c08ab13637c..5c7597191e3f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -774,7 +774,7 @@ static int fsl_esai_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return irq; } diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f4a10d..5e793bbb6b02 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 032d2d33619c..ec79c3d5e65e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -612,7 +612,7 @@ static int fsl_sai_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return irq; } diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index af0429421fc8..75870c0ea2c9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -90,7 +90,6 @@ struct spdif_mixer_control { * @sysclk: system clock for rx clock rate measurement * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel - * @name: driver name */ struct fsl_spdif_priv { struct spdif_mixer_control fsl_spdif_control; @@ -109,12 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; - - /* The name space will be allocated dynamically */ - char name[0]; }; - /* DPLL locked and lock loss interrupt handler */ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) { @@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev) if (!np) return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, - sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, - GFP_KERNEL); + spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; - strcpy(spdif_priv->name, np->name); - spdif_priv->pdev = pdev; /* Initialize this copy of the CPU DAI driver structure */ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); - spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1198,12 +1189,12 @@ static int fsl_spdif_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return irq; } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, - spdif_priv->name, spdif_priv); + dev_name(&pdev->dev), spdif_priv); if (ret) { dev_err(&pdev->dev, "could not claim irq %u\n", irq); return ret; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d57ffb..46549de60e50 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -160,7 +160,7 @@ struct fsl_ssi_soc_data { */ struct fsl_ssi_private { struct regmap *regs; - unsigned int irq; + int irq; struct snd_soc_dai_driver cpu_dai_drv; unsigned int dai_fmt; @@ -1363,8 +1363,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - return -ENXIO; + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + return ssi_private->irq; } /* Are the RX and the TX clocks locked? */ diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e94704f1b9ee..33da26a12457 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; data->card.dai_link = &data->dai; data->card.num_links = 1; + data->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 4caacb05a623..cd146d4fa805 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) if (ret) goto clk_fail; data->card.num_links = 1; + data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_wm8962_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index b1ced7b8d80c..198eeb3f3f7a 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -55,16 +55,6 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - u32 dai_format; - - dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 25000000, SND_SOC_CLOCK_OUT); @@ -164,6 +154,8 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .platform_name = "imx-ssi.0", .codec_name = "tlv320aic32x4.0-0018", .cpu_dai_name = "imx-ssi.0", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &mx27vis_aic32x4_snd_ops, }; diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 804749a6c61e..d072bd13db09 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -87,7 +87,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int rate = params_rate(params); unsigned int channels = params_channels(params); - u32 dai_format; /* find the correct audio parameters */ for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { @@ -104,15 +103,6 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, /* codec FLL input is 14.75 MHz from MCLK */ snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); - dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); - /* TODO: The SSI driver should figure this out for us */ switch (channels) { case 2: @@ -244,6 +234,8 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, .symmetric_rates = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card wm1133_ev1 = { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240fdc9b7..7fe3009b1c43 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Decrease the reference count of the device nodes */ -static int asoc_simple_card_unref(struct platform_device *pdev) +static int asoc_simple_card_unref(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); struct snd_soc_dai_link *dai_link; int num_links; @@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return ret; err: - asoc_simple_card_unref(pdev); + asoc_simple_card_unref(&priv->snd_card); return ret; } @@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); - return asoc_simple_card_unref(pdev); + return asoc_simple_card_unref(card); } static const struct of_device_id asoc_simple_of_match[] = { diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf046c9..ee03dbdda235 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -46,7 +46,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 @@ -76,7 +76,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD @@ -89,7 +89,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 + depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -101,7 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH platforms with RT5672 audio codec. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5645_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5645 audio codec. + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index e928ec385300..a8e53c45c6b6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 7cf95d5d5d80..9cf7d01479ad 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -140,8 +140,6 @@ static struct snd_soc_ops broadwell_rt286_ops = { static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); struct sst_hsw *broadwell = pdata->dsp; int ret; @@ -155,14 +153,6 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "DMIC1"); - snd_soc_dapm_enable_pin(dapm, "DMIC2"); - return 0; } diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 0cba7830c5e9..354eaad886e1 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -132,7 +132,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; @@ -161,7 +160,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); } - ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); + ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); if (ret) return ret; @@ -171,13 +170,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } - snd_soc_dapm_ignore_suspend(dapm, "HPOL"); - snd_soc_dapm_ignore_suspend(dapm, "HPOR"); - - snd_soc_dapm_ignore_suspend(dapm, "SPOLP"); - snd_soc_dapm_ignore_suspend(dapm, "SPOLN"); - snd_soc_dapm_ignore_suspend(dapm, "SPORP"); - snd_soc_dapm_ignore_suspend(dapm, "SPORN"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); return ret; } diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index f5d0fc1ab10c..59308629043e 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -215,7 +215,6 @@ static int snd_byt_mc_probe(struct platform_device *pdev) static struct platform_driver snd_byt_mc_driver = { .driver = { - .owner = THIS_MODULE, .name = "bytt100_rt5640", .pm = &snd_soc_pm_ops, }, @@ -227,4 +226,4 @@ module_platform_driver(snd_byt_mc_driver); MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bytrt5640-audio"); +MODULE_ALIAS("platform:bytt100_rt5640"); diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c new file mode 100644 index 000000000000..bd29617a9ab9 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -0,0 +1,326 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2015 Intel Corp + * Author: Fang, Yang A <yang.a.fang@intel.com> + * N,Harshapriya <harshapriya.n@intel.com> + * This file is modified from cht_bsw_rt5672.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" +#include "sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + + /* Select clk_i2s1_asrc as ASRC clock source */ + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, + &ctx->hp_jack); + if (ret) { + dev_err(runtime->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Mic Jack", + SND_JACK_MICROPHONE, + &ctx->mic_jack); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 9b8b561171b7..ff016621583a 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -140,6 +140,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); @@ -148,6 +149,19 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } + /* Select codec ASRC clock source to track I2S1 clock, because codec + * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot + * be supported by RT5672. Otherwise, ASRC will be disabled and cause + * noise. + */ + rt5670_sel_asrc_clk_src(codec, + RT5670_DA_STEREO_FILTER + | RT5670_DA_MONO_L_FILTER + | RT5670_DA_MONO_R_FILTER + | RT5670_AD_STEREO_FILTER + | RT5670_AD_MONO_L_FILTER + | RT5670_AD_MONO_R_FILTER, + RT5670_CLK_SEL_I2S1_ASRC); return 0; } @@ -270,7 +284,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { - .owner = THIS_MODULE, .name = "cht-bsw-rt5672", .pm = &snd_soc_pm_ops, }, diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 3bb6288d8b4d..224c49c9f135 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -320,11 +320,6 @@ static struct snd_pcm_ops sst_byt_pcm_ops = { .mmap = sst_byt_pcm_mmap, }; -static void sst_byt_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; @@ -403,7 +398,6 @@ static struct snd_soc_platform_driver byt_soc_platform = { .remove = sst_byt_pcm_remove, .ops = &sst_byt_pcm_ops, .pcm_new = sst_byt_pcm_new, - .pcm_free = sst_byt_pcm_free, }; static const struct snd_soc_component_driver byt_dai_component = { diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 86e410845670..64e94212d2d2 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -410,8 +410,7 @@ void sst_dsp_free(struct sst_dsp *sst) if (sst->ops->free) sst->ops->free(sst); - if (sst->dma) - sst_dma_free(sst->dma); + sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 4a5bde9c686b..5f71ef607a57 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -497,6 +497,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->sst_fw = sst_fw; sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; + sst_module->entry = template->entry; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); @@ -706,6 +707,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba struct list_head *block_list) { struct sst_mem_block *block, *tmp; + struct sst_block_allocator ba_tmp = *ba; u32 end = ba->offset + ba->size, block_end; int err; @@ -730,9 +732,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba if (ba->offset >= block->offset && ba->offset < block_end) { /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -763,10 +765,14 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba /* does block span more than 1 section */ if (ba->offset >= block->offset && ba->offset < block_end) { + /* add block */ + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->offset = block->offset; + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -785,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module) struct sst_block_allocator ba; int ret; + memset(&ba, 0, sizeof(ba)); ba.size = module->size; ba.type = module->type; ba.offset = module->offset; @@ -858,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, if (module->persistent_size == 0) return 0; + memset(&ba, 0, sizeof(ba)); ba.size = module->persistent_size; ba.type = SST_MEM_DRAM; diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 57039b00efc2..c42ffae5fe9f 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -306,7 +306,7 @@ static void hsw_reset(struct sst_dsp *sst) static int hsw_set_dsp_D0(struct sst_dsp *sst) { int tries = 10; - u32 reg; + u32 reg, fw_dump_bit; /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); @@ -368,7 +368,9 @@ finish: can't be accessed, please enable each block before accessing. */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* disable DMA finish function for SSP0 & SSP1 */ @@ -491,6 +493,7 @@ static const struct sst_sram_shift sram_shift[] = { {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ }; + static u32 hsw_block_get_bit(struct sst_mem_block *block) { u32 bit = 0, shift = 0, index; @@ -587,7 +590,9 @@ static int hsw_block_disable(struct sst_mem_block *block) val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); - writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* don't disable DSRAM[0], keep it always enable for FW dump*/ + if (bit != (1 << SST_VDRTCL0_DSRAMPGE_SHIFT)) + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* wait 18 DSP clock ticks */ udelay(10); @@ -612,7 +617,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) const struct sst_adsp_memregion *region; struct device *dev; int ret = -ENODEV, i, j, region_count; - u32 offset, size; + u32 offset, size, fw_dump_bit; dev = sst->dma_dev; @@ -669,9 +674,11 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } + /* always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; /* set default power gating control, enable power gating control for all blocks. that is, can't be accessed, please enable each block before accessing. */ - writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); + writel(0xffffffff & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c48231364..0ab1309ef274 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -94,6 +94,8 @@ /* Mailbox */ #define IPC_MAX_MAILBOX_BYTES 256 +#define INVALID_STREAM_HW_ID 0xffffffff + /* Global Message - Types and Replies */ enum ipc_glb_type { IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ @@ -275,7 +277,6 @@ struct sst_hsw { /* FW config */ struct sst_hsw_ipc_fw_ready fw_ready; struct sst_hsw_ipc_fw_version version; - struct sst_module *scratch; bool fw_done; struct sst_fw *sst_fw; @@ -337,12 +338,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -651,11 +646,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) @@ -969,45 +964,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1021,17 +977,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1028,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1041,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1208,6 +1107,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, return NULL; spin_lock_irqsave(&sst->spinlock, flags); + stream->reply.stream_hw_id = INVALID_STREAM_HW_ID; list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; @@ -1228,6 +1128,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) struct sst_dsp *sst = hsw->dsp; unsigned long flags; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n"); + return 0; + } + /* dont free DSP streams that are not commited */ if (!stream->commited) goto out; @@ -1415,6 +1320,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) u32 header; int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n"); + return 0; + } + + if (stream->commited) { + dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n"); + return 0; + } + trace_ipc_request("stream alloc", stream->host_id); header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); @@ -1434,48 +1349,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1519,6 +1392,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n"); + return 0; + } + trace_ipc_request("stream pause", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, @@ -1535,6 +1413,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n"); + return 0; + } + trace_ipc_request("stream resume", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, @@ -1550,6 +1433,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { int ret, tries = 10; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n"); + return 0; + } + /* dont reset streams that are not commited */ if (!stream->commited) return 0; @@ -1598,30 +1486,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, @@ -2102,7 +1966,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kfree(hsw->scratch); kthread_stop(hsw->tx_thread); kfree(hsw->msg); } diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894ab413..c1ad901342f2 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 619525200705..78fa01be57f2 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -78,7 +78,6 @@ static const u32 volume_map[] = { #define HSW_PCM_DAI_ID_OFFLOAD0 1 #define HSW_PCM_DAI_ID_OFFLOAD1 2 #define HSW_PCM_DAI_ID_LOOPBACK 3 -#define HSW_PCM_DAI_ID_CAPTURE 4 static const struct snd_pcm_hardware hsw_pcm_hardware = { @@ -99,6 +98,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { struct hsw_pcm_module_map { int dai_id; + int stream; enum sst_hsw_module_id mod_id; }; @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -135,7 +136,17 @@ struct hsw_priv_data { struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; /* DAI data */ - struct hsw_pcm_data pcm[HSW_PCM_COUNT]; + struct hsw_pcm_data pcm[HSW_PCM_COUNT][2]; +}; + + +/* static mappings between PCMs and modules - may be dynamic in future */ +static struct hsw_pcm_module_map mod_map[] = { + {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, }; static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data); @@ -168,9 +179,14 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -212,9 +228,14 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -309,7 +330,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -353,7 +374,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; struct sst_module *module_data; struct sst_dsp *dsp; @@ -362,7 +383,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, enum sst_hsw_stream_path_id path_id; u32 rate, bits, map, pages, module_id; u8 channels; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ if (pcm_data->allocated) { @@ -552,8 +576,12 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,11 +625,16 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; snd_pcm_uframes_t offset; uint64_t ppos; - u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); + u32 position; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; + position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); @@ -618,8 +651,10 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; - pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -648,9 +683,12 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); ret = sst_hsw_stream_reset(hsw, pcm_data->stream); @@ -685,15 +723,6 @@ static struct snd_pcm_ops hsw_pcm_ops = { .page = snd_pcm_sgbuf_ops_page, }; -/* static mappings between PCMs and modules - may be dynamic in future */ -static struct hsw_pcm_module_map mod_map[] = { - {HSW_PCM_DAI_ID_SYSTEM, SST_HSW_MODULE_PCM_SYSTEM}, - {HSW_PCM_DAI_ID_OFFLOAD0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_OFFLOAD1, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_LOOPBACK, SST_HSW_MODULE_PCM_REFERENCE}, - {HSW_PCM_DAI_ID_CAPTURE, SST_HSW_MODULE_PCM_CAPTURE}, -}; - static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) { struct sst_hsw *hsw = pdata->hsw; @@ -701,7 +730,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; /* create new runtime module, use same offset if recreated */ pcm_data->runtime = sst_hsw_runtime_module_create(hsw, @@ -716,7 +745,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) err: for (--i; i >= 0; i--) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -729,17 +758,12 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } } -static void hsw_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; @@ -762,7 +786,10 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } } - priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return ret; } @@ -871,10 +898,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { - mutex_init(&priv_data->pcm[i].mutex); - /* playback */ if (hsw_dais[i].playback.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_PLAYBACK].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][0]); if (ret < 0) @@ -883,6 +909,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* capture */ if (hsw_dais[i].capture.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_CAPTURE].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][1]); if (ret < 0) @@ -936,7 +963,6 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .remove = hsw_pcm_remove, .ops = &hsw_pcm_ops, .pcm_new = hsw_pcm_new, - .pcm_free = hsw_pcm_free, }; static const struct snd_soc_component_driver hsw_dai_component = { @@ -1010,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1026,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1066,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1081,8 +1107,8 @@ static void hsw_pcm_complete(struct device *dev) return; } - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1114,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; + else if (pdata->pm_state == HSW_PM_STATE_D0) { + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index a1a8d9d91539..7523cbef8780 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -643,12 +643,6 @@ static struct snd_pcm_ops sst_platform_ops = { .pointer = sst_platform_pcm_pointer, }; -static void sst_pcm_free(struct snd_pcm *pcm) -{ - dev_dbg(pcm->dev, "sst_pcm_free called\n"); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; @@ -679,7 +673,6 @@ static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, - .pcm_free = sst_pcm_free, }; static const struct snd_soc_component_driver sst_component = { diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 7f4bbfcbc6f5..562bc483d6b7 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -58,6 +58,7 @@ enum sst_algo_ops { #define SST_BLOCK_TIMEOUT 1000 #define FW_SIGNATURE_SIZE 4 +#define FW_NAME_SIZE 32 /* stream states */ enum sst_stream_states { @@ -426,7 +427,7 @@ struct intel_sst_drv { * Holder for firmware name. Due to async call it needs to be * persistent till worker thread gets called */ - char firmware_name[20]; + char firmware_name[FW_NAME_SIZE]; }; /* misc definitions */ diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3abc29e8a928..b782dfdcdbba 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -47,7 +47,7 @@ struct sst_machines { char board[32]; char machine[32]; void (*machine_quirk)(void); - char firmware[32]; + char firmware[FW_NAME_SIZE]; struct sst_platform_info *pdata; }; @@ -245,7 +245,7 @@ static struct sst_machines *sst_acpi_find_machine( return NULL; } -int sst_acpi_probe(struct platform_device *pdev) +static int sst_acpi_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; int ret = 0; @@ -332,7 +332,7 @@ do_sst_cleanup: * This function is called by OS when a device is unloaded * This frees the interrupt etc */ -int sst_acpi_remove(struct platform_device *pdev) +static int sst_acpi_remove(struct platform_device *pdev) { struct intel_sst_drv *ctx; @@ -343,14 +343,16 @@ int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; /* Cherryview-based platforms: CherryTrail and Braswell */ static struct sst_machines sst_acpi_chv[] = { - {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", + &chv_platform_data }, + {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; @@ -366,7 +368,6 @@ MODULE_DEVICE_TABLE(acpi, sst_acpi_ids); static struct platform_driver sst_acpi_driver = { .driver = { .name = "intel_sst_acpi", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(sst_acpi_ids), .pm = &intel_sst_pm, }, diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index b580f96e25e5..7888cd707853 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -324,8 +324,7 @@ void sst_firmware_load_cb(const struct firmware *fw, void *context) if (ctx->sst_state != SST_RESET || ctx->fw_in_mem != NULL) { - if (fw != NULL) - release_firmware(fw); + release_firmware(fw); mutex_unlock(&ctx->sst_lock); return; } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index d3d45c6f064f..07f77815a586 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -14,6 +14,8 @@ #include <linux/init.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -83,6 +85,8 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf +#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) struct jz4740_i2s { struct resource *mem; @@ -237,10 +241,14 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int sample_size; - uint32_t ctrl; + uint32_t ctrl, div_reg; + int div; ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + div_reg = jz4740_i2s_read(i2s, JZ_REG_AIC_CLK_DIV); + div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: sample_size = 0; @@ -264,7 +272,10 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; } + div_reg &= ~I2SDIV_DV_MASK; + div_reg |= (div - 1) << I2SDIV_DV_SHIFT; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg); return 0; } @@ -415,6 +426,13 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", }; +#ifdef CONFIG_OF +static const struct of_device_id jz4740_of_matches[] = { + { .compatible = "ingenic,jz4740-i2s" }, + { /* sentinel */ } +}; +#endif + static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; @@ -455,6 +473,7 @@ static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, .driver = { .name = "jz4740-i2s", + .of_match_table = of_match_ptr(jz4740_of_matches) }, }; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 6f1916b71815..6e6fce6a14ba 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -36,7 +36,7 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate = params_rate(params); - u32 dai_format, mclk; + u32 mclk; int ret; /* sgtl5000 does not support 512*rate when in 96000 fs */ @@ -65,26 +65,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, return ret; } - /* set codec to slave mode */ - dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, dai_format); - if (ret) { - dev_err(codec_dai->dev, "Failed to set dai format to %08x\n", - dai_format); - return ret; - } - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); - if (ret) { - dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n", - dai_format); - return ret; - } - return 0; } @@ -92,17 +72,22 @@ static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { .hw_params = mxs_sgtl5000_hw_params, }; +#define MXS_SGTL5000_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS) + static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { { .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", + .dai_fmt = MXS_SGTL5000_DAI_FMT, .ops = &mxs_sgtl5000_hifi_ops, .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", + .dai_fmt = MXS_SGTL5000_DAI_FMT, .ops = &mxs_sgtl5000_hifi_ops, .capture_only = true, }, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 4c6afb75eea6..706613077c15 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -412,21 +412,7 @@ static struct tty_ldisc_ops cx81801_ops = { * over the modem port. */ -static int ams_delta_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* Set cpu DAI configuration */ - return snd_soc_dai_set_fmt(rtd->cpu_dai, - SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -} - -static struct snd_soc_ops ams_delta_ops = { - .hw_params = ams_delta_hw_params, -}; +static struct snd_soc_ops ams_delta_ops; /* Digital mute implemented using modem/CPU multiplexer. @@ -546,6 +532,8 @@ static struct snd_soc_dai_link ams_delta_dai_link = { .platform_name = "omap-mcbsp.1", .codec_name = "cx20442-codec", .ops = &ams_delta_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; /* Audio card driver */ diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index 3f9ac7dbdc80..ccfb41c22e53 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -393,7 +393,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev) static struct platform_driver hdmi_audio_driver = { .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, }, .probe = omap_hdmi_audio_probe, .remove = omap_hdmi_audio_remove, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8b79cafab1e2..c7eb9dd67f60 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBM_CFS: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; - regs->pcr0 |= FSXM; + regs->pcr0 |= FSXM | FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 5e551c762b7a..fb1f6bb87cd4 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -53,11 +53,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_card *card = rtd->card; unsigned int fmt; - int ret; switch (params_channels(params)) { case 2: /* Stereo I2S mode */ @@ -74,21 +70,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - dev_err(card->dev, "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - dev_err(card->dev, "can't set cpu DAI configuration\n"); - return ret; - } - - return 0; + return snd_soc_runtime_set_dai_fmt(rtd, fmt); } static struct snd_soc_ops omap_twl4030_ops = { diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 396dbd51a64f..a9615a574546 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -81,7 +81,7 @@ static int rear_amp_power(struct snd_soc_codec *codec, int power) static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kctl, int event) { - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec; return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); } diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 083706595495..552b763005ed 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -88,7 +88,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int fmt, clk = 0; + unsigned int clk = 0; int ret = 0; switch (params_rate(params)) { @@ -112,15 +112,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - /* setup the CODEC DAI */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); if (ret < 0) return ret; @@ -130,10 +121,6 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); if (ret < 0) return ret; @@ -169,9 +156,8 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int fmt, ret = 0, clk = 0; + int ret = 0, clk = 0; switch (params_rate(params)) { case 44100: @@ -194,22 +180,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; - - /* setup the CODEC DAI */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* setup the CPU DAI */ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); if (ret < 0) return ret; @@ -233,6 +208,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ .codec_name = "cs4270.0-0048", \ + .dai_fmt = SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS, \ .ops = &raumfeld_cs4270_ops, \ } @@ -243,6 +221,9 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .cpu_dai_name = "pxa-ssp-dai.1", \ .codec_dai_name = "ak4104-hifi", \ .platform_name = "pxa-pcm-audio", \ + .dai_fmt = SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBS_CFS, \ .ops = &raumfeld_ak4104_ops, \ .codec_name = "spi0.0", \ } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d7d5fb20ea6f..a6d680acd907 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -352,7 +352,6 @@ static int spitz_remove(struct platform_device *pdev) static struct platform_driver spitz_driver = { .driver = { .name = "spitz-audio", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, }, .probe = spitz_probe, diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 23bf991e95d5..8f301c72ee5e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -130,16 +130,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - return 0; } @@ -172,6 +162,8 @@ static struct snd_soc_dai_link zylonite_dai[] = { .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", .codec_dai_name = "wm9713-voice", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &zylonite_voice_ops, }, }; diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 26ec5117b35c..acb5be53bfb4 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -247,6 +247,10 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, + I2S_DMACR_TDL(16)); + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, + I2S_DMACR_RDL(16)); return 0; } @@ -335,6 +339,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, }; static const struct snd_soc_component_driver rockchip_i2s_component = { @@ -454,11 +459,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->playback_dma_data.addr = res->start + I2S_TXDR; i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->playback_dma_data.maxburst = 16; + i2s->playback_dma_data.maxburst = 4; i2s->capture_dma_data.addr = res->start + I2S_RXDR; i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->capture_dma_data.maxburst = 16; + i2s->capture_dma_data.maxburst = 4; i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index 89a5d8bc6ee7..93f456f518a9 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -127,7 +127,7 @@ #define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT) #define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT) #define I2S_DMACR_TDL_SHIFT 0 -#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT) +#define I2S_DMACR_TDL(x) ((x) << I2S_DMACR_TDL_SHIFT) #define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT) /* diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 1e2b61ca8db2..8bf2e2c4bafb 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -135,7 +135,6 @@ MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match); static struct platform_driver arndale_audio_driver = { .driver = { .name = "arndale-audio", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 3b527dcfc0aa..fad56b9e7369 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -136,22 +136,9 @@ static int goni_hifi_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int pll_out = 24000000; int ret = 0; - /* set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, params_rate(params) * 256); @@ -182,12 +169,6 @@ static int goni_voice_hw_params(struct snd_pcm_substream *substream, if (params_rate(params) != 8000) return -EINVAL; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, params_rate(params) * 256); @@ -234,6 +215,8 @@ static struct snd_soc_dai_link goni_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-i2s.0", .codec_name = "wm8994-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = goni_wm8994_init, .ops = &goni_hifi_ops, }, { @@ -242,6 +225,8 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-aif2", .codec_name = "wm8994-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &goni_voice_ops, }, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index f2d7980d7ddc..59b044255b78 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -76,7 +76,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int div; int ret; unsigned int rate = params_rate(params); @@ -95,18 +94,6 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* select clock source */ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, SND_SOC_CLOCK_OUT); @@ -207,6 +194,8 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { .init = h1940_uda1380_init, .platform_name = "s3c24xx-iis", .codec_name = "uda1380-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &h1940_ops, }, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index b5f6abd9d221..6c3b359bb4c1 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -61,20 +61,6 @@ static int jive_hw_params(struct snd_pcm_substream *substream, s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), s3c_i2sv2_get_clock(cpu_dai)); - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); @@ -121,6 +107,8 @@ static struct snd_soc_dai_link jive_dai = { .platform_name = "s3c2412-i2s", .codec_name = "wm8750.0-001a", .init = jive_wm8750_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 9b4a09f14b6c..65602b935377 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -70,20 +70,6 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); @@ -151,13 +137,6 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); @@ -300,6 +279,8 @@ static struct snd_soc_dai_link neo1973_dai[] = { .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .codec_name = "wm8753.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -309,6 +290,8 @@ static struct snd_soc_dai_link neo1973_dai[] = { .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 37688ebbb2b4..873f2cb4bebe 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -89,6 +89,8 @@ static struct snd_soc_dai_link rx1950_uda1380_dai[] = { .init = rx1950_uda1380_init, .platform_name = "s3c24xx-iis", .codec_name = "uda1380-codec.0-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &rx1950_ops, }, }; @@ -154,7 +156,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int div; int ret; unsigned int rate = params_rate(params); @@ -181,18 +182,6 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* select clock source */ ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source, rate, SND_SOC_CLOCK_OUT); diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 2c015f62ead6..dcc008d1e1ab 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -169,24 +169,6 @@ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - /* Set the CODEC as the bus clock master, I2S */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - /* Set the CODEC as the bus clock master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { @@ -320,6 +302,8 @@ int simtec_audio_core_probe(struct platform_device *pdev, int ret; card->dai_link->ops = &simtec_snd_ops; + card->dai_link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; pdata = pdev->dev.platform_data; if (!pdata) { diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 9c6f7db56f60..50849e137fc0 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -173,16 +173,6 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -223,6 +213,8 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &s3c24xx_uda134x_ops, .platform_name = "s3c24xx-iis", }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 9b0ffacab790..8291d2a5f152 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -56,20 +56,6 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* Use PCLK for I2S signal generation */ ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, 0, SND_SOC_CLOCK_IN); @@ -199,6 +185,8 @@ static struct snd_soc_dai_link smartq_dai[] = { .platform_name = "samsung-i2s.0", .codec_name = "wm8750.0-0x1a", .init = smartq_wm8987_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &smartq_hifi_ops, }, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index b1a519f83b29..17a2f717ec02 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -32,7 +32,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int pll_out; int bfs, rfs, ret; @@ -77,20 +76,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, } pll_out = params_rate(params) * rfs; - /* Set the Codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* Set the AP DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* Set WM8580 to drive MCLK from its PLLA */ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, WM8580_CLKSRC_PLLA); @@ -168,6 +153,9 @@ enum { SEC_PLAYBACK, }; +#define SMDK_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link smdk_dai[] = { [PRI_PLAYBACK] = { /* Primary Playback i/f */ .name = "WM8580 PAIF RX", @@ -176,6 +164,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-i2s.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_ops, }, [PRI_CAPTURE] = { /* Primary Capture i/f */ @@ -185,6 +174,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-i2s.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .init = smdk_wm8580_init_paiftx, .ops = &smdk_ops, }, @@ -195,6 +185,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-i2s-sec", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_ops, }, }; diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 05c609c62de9..6deec5234c92 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -62,20 +62,6 @@ static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream, rfs = mclk_freq / params_rate(params) / 2; - /* Set the codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* Set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - if (mclk_freq == xtal_freq) { ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, mclk_freq, SND_SOC_CLOCK_IN); @@ -121,6 +107,9 @@ static struct snd_soc_ops smdk_wm8580_pcm_ops = { .hw_params = smdk_wm8580_pcm_hw_params, }; +#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \ + SND_SOC_DAIFMT_CBS_CFS) + static struct snd_soc_dai_link smdk_dai[] = { { .name = "WM8580 PAIF PCM RX", @@ -129,6 +118,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-playback", .platform_name = "samsung-audio", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_wm8580_pcm_ops, }, { .name = "WM8580 PAIF PCM TX", @@ -137,6 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8580-hifi-capture", .platform_name = "samsung-pcm.0", .codec_name = "wm8580.0-001b", + .dai_fmt = SMDK_DAI_FMT, .ops = &smdk_wm8580_pcm_ops, }, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index c470e8eed6e1..b1c89ec2d999 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -68,20 +68,6 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, mclk_freq = params_rate(params) * rfs; - /* Set the codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* Set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B - | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, mclk_freq, SND_SOC_CLOCK_IN); if (ret < 0) @@ -118,6 +104,8 @@ static struct snd_soc_dai_link smdk_dai[] = { .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-pcm.0", .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &smdk_wm8994_pcm_ops, }, }; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8869971d7884..d49f25f9efd3 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -820,12 +820,9 @@ static int fsi_clk_enable(struct device *dev, return ret; } - if (clock->xck) - clk_enable(clock->xck); - if (clock->ick) - clk_enable(clock->ick); - if (clock->div) - clk_enable(clock->div); + clk_enable(clock->xck); + clk_enable(clock->ick); + clk_enable(clock->div); clock->count++; } diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index c58c2529f103..82f582344fe7 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -63,16 +63,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - codec_freq = rate * 512; /* * This propagates the parent frequency change to children and @@ -144,6 +134,8 @@ static struct snd_soc_dai_link migor_dai = { .codec_dai_name = "wm8978-hifi", .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", + .dai_fmt = SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFS, .ops = &migor_dai_ops, }; diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 2e10e9a38376..08d7259bbaab 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -48,15 +48,18 @@ static void soc_ac97_device_release(struct device *dev) } /** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec + * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device + * @codec: The CODEC for which to create the AC'97 device * - * Initialises AC97 codec resources for use by ad-hoc devices only. + * Allocated a new snd_ac97 device and intializes it, but does not yet register + * it. The caller is responsible to either call device_add(&ac97->dev) to + * register the device, or to call put_device(&ac97->dev) to free the device. + * + * Returns: A snd_ac97 device or a PTR_ERR in case of an error. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret; ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (ac97 == NULL) @@ -73,7 +76,28 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) codec->component.card->snd_card->number, 0, codec->component.name); - ret = device_register(&ac97->dev); + device_initialize(&ac97->dev); + + return ac97; +} +EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); + +/** + * snd_soc_new_ac97_codec - initailise AC97 device + * @codec: audio codec + * + * Initialises AC97 codec resources for use by ad-hoc devices only. + */ +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97; + int ret; + + ac97 = snd_soc_alloc_ac97_codec(codec); + if (IS_ERR(ac97)) + return ac97; + + ret = device_add(&ac97->dev); if (ret) { put_device(&ac97->dev); return ERR_PTR(ret); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 985052b3fbed..678823d2e14a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static umode_t soc_dev_attr_is_visible(struct kobject *kobj, + struct attribute *attr, int idx) +{ + struct device *dev = kobj_to_dev(kobj); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + + if (attr == &dev_attr_pmdown_time.attr) + return attr->mode; /* always visible */ + return rtd->codec ? attr->mode : 0; /* enabled only with codec */ +} + +static const struct attribute_group soc_dapm_dev_group = { + .attrs = soc_dapm_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group soc_dev_roup = { + .attrs = soc_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dapm_dev_group, + &soc_dev_roup, + NULL +}; + #ifdef CONFIG_DEBUG_FS static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); @@ -1427,11 +1437,76 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) return 0; } +/** + * snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime + * @rtd: The runtime for which the DAI link format should be changed + * @dai_fmt: The new DAI link format + * + * This function updates the DAI link format for all DAIs connected to the DAI + * link for the specified runtime. + * + * Note: For setups with a static format set the dai_fmt field in the + * corresponding snd_dai_link struct instead of using this function. + * + * Returns 0 on success, otherwise a negative error code. + */ +int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, + unsigned int dai_fmt) +{ + struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int i; + int ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = codec_dais[i]; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(codec_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + } + + /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ + if (cpu_dai->codec) { + unsigned int inv_dai_fmt; + + inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; + } + + dai_fmt = inv_dai_fmt; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(cpu_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; - struct snd_soc_dai_link *dai_link; - int ret, i, order, dai_fmt; + int ret, i, order; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); @@ -1542,60 +1617,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); for (i = 0; i < card->num_links; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - dai_link = &card->dai_link[i]; - dai_fmt = dai_link->dai_fmt; - - if (dai_fmt) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = codec_dais[j]; - - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(codec_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } - } - - /* If this is a regular CPU link there will be a platform */ - if (dai_fmt && - (dai_link->platform_name || dai_link->platform_of_node)) { - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } else if (dai_fmt) { - /* Flip the polarity for the "CPU" end */ - dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - switch (dai_link->dai_fmt & - SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case SND_SOC_DAIFMT_CBM_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case SND_SOC_DAIFMT_CBS_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } + if (card->dai_link[i].dai_fmt) + snd_soc_runtime_set_dai_fmt(&card->rtd[i], + card->dai_link[i].dai_fmt); } snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), @@ -1626,9 +1650,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - if (card->fully_routed) - snd_soc_dapm_auto_nc_pins(card); - snd_soc_dapm_new_widgets(card); ret = snd_card_register(card->snd_card); @@ -2386,8 +2407,8 @@ int snd_soc_unregister_card(struct snd_soc_card *card) card->instantiated = false; snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3230,7 +3251,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes, old_routes; + int num_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -3248,9 +3269,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - old_routes = card->num_dapm_routes; - routes = devm_kzalloc(card->dev, - (old_routes + num_routes) * sizeof(*routes), + routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -3258,11 +3277,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); - for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[old_routes + i].sink); + 2 * i, &routes[i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3270,7 +3287,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[old_routes + i].source); + (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3279,7 +3296,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes += num_routes; + card->num_dapm_routes = num_routes; card->dapm_routes = routes; return 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..b6f88202b8c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -517,8 +517,8 @@ static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm, { if (!dapm->component) return -EIO; - return snd_soc_component_update_bits_async(dapm->component, reg, - mask, value); + return snd_soc_component_update_bits(dapm->component, reg, + mask, value); } static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm, @@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} - -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -2279,6 +2274,9 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) switch (w->id) { case snd_soc_dapm_input: + /* On a fully routed card a input is never a source */ + if (w->dapm->card->fully_routed) + break; w->is_source = 1; list_for_each_entry(p, &w->sources, list_sink) { if (p->source->id == snd_soc_dapm_micbias || @@ -2291,6 +2289,9 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) } break; case snd_soc_dapm_output: + /* On a fully routed card a output is never a sink */ + if (w->dapm->card->fully_routed) + break; w->is_sink = 1; list_for_each_entry(p, &w->sinks, list_source) { if (p->sink->id == snd_soc_dapm_spk || @@ -3085,16 +3086,24 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_mic: - case snd_soc_dapm_input: w->is_source = 1; w->power_check = dapm_generic_check_power; break; + case snd_soc_dapm_input: + if (!dapm->card->fully_routed) + w->is_source = 1; + w->power_check = dapm_generic_check_power; + break; case snd_soc_dapm_spk: case snd_soc_dapm_hp: - case snd_soc_dapm_output: w->is_sink = 1; w->power_check = dapm_generic_check_power; break; + case snd_soc_dapm_output: + if (!dapm->card->fully_routed) + w->is_sink = 1; + w->power_check = dapm_generic_check_power; + break; case snd_soc_dapm_vmid: case snd_soc_dapm_siggen: w->is_source = 1; @@ -3130,8 +3139,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - if (dapm->component) - w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); @@ -3809,93 +3816,6 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** - * dapm_is_external_path() - Checks if a path is a external path - * @card: The card the path belongs to - * @path: The path to check - * - * Returns true if the path is either between two different DAPM contexts or - * between two external pins of the same DAPM context. Otherwise returns - * false. - */ -static bool dapm_is_external_path(struct snd_soc_card *card, - struct snd_soc_dapm_path *path) -{ - dev_dbg(card->dev, - "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n", - path->source->name, path->source->id, path->source->dapm, - path->sink->name, path->sink->id, path->sink->dapm); - - /* Connection between two different DAPM contexts */ - if (path->source->dapm != path->sink->dapm) - return true; - - /* Loopback connection from external pin to external pin */ - if (path->sink->id == snd_soc_dapm_input) { - switch (path->source->id) { - case snd_soc_dapm_output: - case snd_soc_dapm_micbias: - return true; - default: - break; - } - } - - return false; -} - -static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, - struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, &w->sources, list_sink) { - if (dapm_is_external_path(card, p)) - return true; - } - - list_for_each_entry(p, &w->sinks, list_source) { - if (dapm_is_external_path(card, p)) - return true; - } - - return false; -} - -/** - * snd_soc_dapm_auto_nc_pins - call snd_soc_dapm_nc_pin for unused pins - * @card: The card whose pins should be processed - * - * Automatically call snd_soc_dapm_nc_pin() for any external pins in the card - * which are unused. Pins are used if they are connected externally to a - * component, whether that be to some other device, or a loop-back connection to - * the component itself. - */ -void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) -{ - struct snd_soc_dapm_widget *w; - - dev_dbg(card->dev, "ASoC: Auto NC: DAPMs: card:%p\n", &card->dapm); - - list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_micbias: - dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n", - w->name); - if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(card->dev, - "... Not in map; disabling\n"); - snd_soc_dapm_nc_pin(w->dapm, w->name); - } - break; - default: - break; - } - } -} - -/** * snd_soc_dapm_free - free dapm resources * @dapm: DAPM context * @@ -3903,7 +3823,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index b329b84bc5af..4864392bfcba 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -200,11 +200,6 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) return snd_dmaengine_pcm_open(substream, chan); } -static void dmaengine_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) @@ -283,8 +278,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!pcm->chan[i]) { dev_err(rtd->platform->dev, "Missing dma channel for stream: %d\n", i); - ret = -EINVAL; - goto err_free; + return -EINVAL; } ret = snd_pcm_lib_preallocate_pages(substream, @@ -293,7 +287,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) prealloc_buffer_size, max_buffer_size); if (ret) - goto err_free; + return ret; /* * This will only return false if we know for sure that at least @@ -307,10 +301,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) } return 0; - -err_free: - dmaengine_pcm_free(rtd->pcm); - return ret; } static snd_pcm_uframes_t dmaengine_pcm_pointer( @@ -341,7 +331,6 @@ static const struct snd_soc_platform_driver dmaengine_pcm_platform = { }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, - .pcm_free = dmaengine_pcm_free, }; static const char * const dmaengine_pcm_dma_channel_names[] = { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb87d96e2cf0..0ae0e2a9eed7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -746,7 +746,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai); if (ret < 0) { dev_err(codec_dai->dev, - "ASoC: DAI prepare error: %d\n", ret); + "ASoC: codec DAI prepare error: %d\n", + ret); goto out; } } @@ -755,8 +756,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); goto out; } } diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index be4f1ac7cd5e..aa65370db82a 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -290,21 +290,9 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_GATED; } - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - dev_err(dev, - "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", - __func__, ret); - return ret; - } - - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - dev_err(dev, - "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", - __func__, ret); + ret = snd_soc_runtime_set_dai_fmt(rtd, fmt); + if (ret) return ret; - } /* Setup TDM-slots */ diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 272844746135..327f8642ca80 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,7 +816,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } - if (cdev->n_streams < 2) { + if (cdev->n_streams < 1) { dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); return -EINVAL; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 41650d5b93b7..3e2ef61c627b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -913,6 +913,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */ case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0819): /* Logitech Webcam C210 */ case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ |