diff options
Diffstat (limited to 'sound')
33 files changed, 235 insertions, 104 deletions
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/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index e5f2fb884bf3..30c673cdc12e 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -188,8 +188,8 @@ 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", }; diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe463102..1d1c7f8a9af2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -861,10 +861,8 @@ 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/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe1464d268..918ada9738b0 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]; @@ -2082,10 +2083,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, 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; } @@ -2100,10 +2105,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, 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; } @@ -2211,9 +2220,11 @@ 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), /* Input Side */ /* micbias */ diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1380d5..75b87c5c0f04 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -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/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/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index b93168d4f648..8d18bbda661b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -209,16 +209,9 @@ 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"); @@ -227,18 +220,22 @@ 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); @@ -263,6 +260,19 @@ 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 +304,7 @@ 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, }; 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_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d57ffb..059496ed9ad7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - return -ENXIO; + return ssi_private->irq; } /* Are the RX and the TX clocks locked? */ 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/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..f86de1211b96 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -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 diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index f5d0fc1ab10c..eef0c56ec32e 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -227,4 +227,4 @@ 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/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 4a5bde9c686b..b3f9489794a6 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -706,6 +706,7 @@ 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 +731,9 @@ 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 +764,14 @@ 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; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c48231364..5bf14040c24a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1228,6 +1228,11 @@ 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 +1420,16 @@ 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); @@ -1519,6 +1534,11 @@ 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 +1555,11 @@ 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 +1575,11 @@ 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; diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3abc29e8a928..2ac72eb5e75d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,7 +343,7 @@ 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 }, {}, }; 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/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 26ec5117b35c..dcc26eda0539 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -335,6 +335,7 @@ 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 +455,11 @@ 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/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..2c62620abca6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3230,7 +3230,7 @@ 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 +3248,7 @@ 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 +3256,9 @@ 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 +3266,7 @@ 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 +3275,7 @@ 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/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 */ |