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-rw-r--r--sound/firewire/amdtp.c71
-rw-r--r--sound/firewire/amdtp.h5
-rw-r--r--sound/firewire/bebob/bebob_stream.c7
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c5
-rw-r--r--sound/firewire/fireworks/fireworks_transaction.c2
-rw-r--r--sound/pci/hda/hda_controller.c24
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/hda_priv.h1
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/soc/adi/axi-i2s.c2
-rw-r--r--sound/soc/codecs/pcm512x.c2
-rw-r--r--sound/soc/codecs/rt286.c6
-rw-r--r--sound/soc/codecs/rt5677.c9
-rw-r--r--sound/soc/dwc/designware_i2s.c49
-rw-r--r--sound/soc/fsl/fsl_esai.h2
-rw-r--r--sound/soc/fsl/fsl_ssi.c4
-rw-r--r--sound/soc/fsl/imx-wm8962.c1
-rw-r--r--sound/soc/omap/omap-mcbsp.c2
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c5
-rw-r--r--sound/soc/rockchip/rockchip_i2s.h2
-rw-r--r--sound/soc/soc-compress.c9
-rw-r--r--sound/soc/soc-core.c14
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/mixer.c1
25 files changed, 158 insertions, 78 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..c0fbe1881439 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];
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/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 */
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-11-28 09:41:41 +0000