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-rw-r--r--sound/firewire/amdtp.c11
-rw-r--r--sound/firewire/amdtp.h1
-rw-r--r--sound/firewire/dice.c29
-rw-r--r--sound/pci/hda/patch_conexant.c9
-rw-r--r--sound/pci/hda/patch_realtek.c2
5 files changed, 41 insertions, 11 deletions
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index f96bf4c7c232..95fc2eaf11dc 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s,
static void update_pcm_pointers(struct amdtp_stream *s,
struct snd_pcm_substream *pcm,
unsigned int frames)
-{ unsigned int ptr;
+{
+ unsigned int ptr;
+
+ /*
+ * In IEC 61883-6, one data block represents one event. In ALSA, one
+ * event equals to one PCM frame. But Dice has a quirk to transfer
+ * two PCM frames in one data block.
+ */
+ if (s->double_pcm_frames)
+ frames *= 2;
ptr = s->pcm_buffer_pointer + frames;
if (ptr >= pcm->runtime->buffer_size)
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index d8ee7b0e9386..4823c08196ac 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -125,6 +125,7 @@ struct amdtp_stream {
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
bool pointer_flush;
+ bool double_pcm_frames;
struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c
index a9a30c0161f1..e3a04d69c853 100644
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
return err;
/*
- * At rates above 96 kHz, pretend that the stream runs at half the
- * actual sample rate with twice the number of channels; two samples
- * of a channel are stored consecutively in the packet. Requires
- * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL.
+ * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in
+ * one data block of AMDTP packet. Thus sampling transfer frequency is
+ * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are
+ * transferred on AMDTP packets at 96 kHz. Two successive samples of a
+ * channel are stored consecutively in the packet. This quirk is called
+ * as 'Dual Wire'.
+ * For this quirk, blocking mode is required and PCM buffer size should
+ * be aligned to SYT_INTERVAL.
*/
channels = params_channels(hw_params);
if (rate_index > 4) {
@@ -579,18 +583,25 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
return err;
}
- for (i = 0; i < channels; i++) {
- dice->stream.pcm_positions[i * 2] = i;
- dice->stream.pcm_positions[i * 2 + 1] = i + channels;
- }
-
rate /= 2;
channels *= 2;
+ dice->stream.double_pcm_frames = true;
+ } else {
+ dice->stream.double_pcm_frames = false;
}
mode = rate_index_to_mode(rate_index);
amdtp_stream_set_parameters(&dice->stream, rate, channels,
dice->rx_midi_ports[mode]);
+ if (rate_index > 4) {
+ channels /= 2;
+
+ for (i = 0; i < channels; i++) {
+ dice->stream.pcm_positions[i] = i * 2;
+ dice->stream.pcm_positions[i + channels] = i * 2 + 1;
+ }
+ }
+
amdtp_stream_set_pcm_format(&dice->stream,
params_format(hw_params));
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6f2fa838b635..6e5d0cb4e3d7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -217,6 +217,7 @@ enum {
CXT_FIXUP_HEADPHONE_MIC_PIN,
CXT_FIXUP_HEADPHONE_MIC,
CXT_FIXUP_GPIO1,
+ CXT_FIXUP_ASPIRE_DMIC,
CXT_FIXUP_THINKPAD_ACPI,
CXT_FIXUP_OLPC_XO,
CXT_FIXUP_CAP_MIX_AMP,
@@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = {
{ }
},
},
+ [CXT_FIXUP_ASPIRE_DMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_stereo_dmic,
+ .chained = true,
+ .chain_id = CXT_FIXUP_GPIO1,
+ },
[CXT_FIXUP_THINKPAD_ACPI] = {
.type = HDA_FIXUP_FUNC,
.v.func = hda_fixup_thinkpad_acpi,
@@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = {
static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC),
SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d446ac3137b3..1ba22fb527c2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -328,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0885:
case 0x10ec0887:
/*case 0x10ec0889:*/ /* this causes an SPDIF problem */
+ case 0x10ec0900:
alc889_coef_init(codec);
break;
case 0x10ec0888:
@@ -2350,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec)
switch (codec->vendor_id) {
case 0x10ec0882:
case 0x10ec0885:
+ case 0x10ec0900:
break;
default:
/* ALC883 and variants */