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sound fixes for 3.17-rc4

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This time it contains a bunch of small ASoC fixes that slipped from in
 previous updates, in addition to the usual HD-audio fixes and the
 regression fixes for FireWire updates in 3.17.
 All commits are reasonably small fixes.
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Merge tag 'sound-3.17-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This time it contains a bunch of small ASoC fixes that slipped from in
  previous updates, in addition to the usual HD-audio fixes and the
  regression fixes for FireWire updates in 3.17.

  All commits are reasonably small fixes"

* tag 'sound-3.17-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix COEF setups for ALC1150 codec
  ASoC: simple-card: Fix bug of wrong decrement DT node's refcount
  ALSA: hda - Fix digital mic on Acer Aspire 3830TG
  ASoC: omap-twl4030: Fix typo in 2nd dai link's platform_name
  ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for Dice quirk
  ALSA: dice: fix wrong channel mappping at higher sampling rate
  ASoC: cs4265: Fix setting of functional mode and clock divider
  ASoC: cs4265: Fix clock rates in clock map table
  ASoC: rt5677: correct mismatch widget name
  ASoC: rt5640: Do not allow regmap to use bulk read-write operations
  ASoC: tegra: Fix typo in include guard
  ASoC: da732x: Fix typo in include guard
  ASoC: core: fix .info for SND_SOC_BYTES_TLV
  ASoC: rcar: Use && instead of & for boolean expressions
  ASoC: Use dev_set_name() instead of init_name
  ASoC: axi: Fix ADI AXI SPDIF specification
This commit is contained in:
Linus Torvalds 2014-09-04 08:49:06 -07:00
commit 57b252f8fd
16 changed files with 67 additions and 28 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
View File

@ -1,7 +1,7 @@
ADI AXI-SPDIF controller
Required properties:
- compatible : Must be "adi,axi-spdif-1.00.a"
- compatible : Must be "adi,axi-spdif-tx-1.00.a"
- reg : Must contain SPDIF core's registers location and length
- clocks : Pairs of phandle and specifier referencing the controller's clocks.
The controller expects two clocks, the clock used for the AXI interface and

2
include/sound/soc.h
View File

@ -277,7 +277,7 @@
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
.tlv.c = (snd_soc_bytes_tlv_callback), \
.info = snd_soc_info_bytes_ext, \
.info = snd_soc_bytes_info_ext, \
.private_value = (unsigned long)&(struct soc_bytes_ext) \
{.max = xcount, .get = xhandler_get, .put = xhandler_put, } }
#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \

11
sound/firewire/amdtp.c
View File

@ -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)

1
sound/firewire/amdtp.h
View File

@ -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];

29
sound/firewire/dice.c
View File

@ -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));

9
sound/pci/hda/patch_conexant.c
View File

@ -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),

2
sound/pci/hda/patch_realtek.c
View File

@ -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 */

12
sound/soc/codecs/cs4265.c
View File

@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
/*64k*/
{8192000, 64000, 1, 0},
{1228800, 64000, 1, 1},
{1693440, 64000, 1, 2},
{2457600, 64000, 1, 3},
{3276800, 64000, 1, 4},
{12288000, 64000, 1, 1},
{16934400, 64000, 1, 2},
{24576000, 64000, 1, 3},
{32768000, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_FM, clk_map_table[index].fm_mode);
CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
clk_map_table[index].mclkdiv);
clk_map_table[index].mclkdiv << 4);
} else {
dev_err(codec->dev, "can't get correct mclk\n");

2
sound/soc/codecs/da732x.h
View File

@ -11,7 +11,7 @@
*/
#ifndef __DA732X_H_
#define __DA732X_H
#define __DA732X_H_
#include <sound/soc.h>

1
sound/soc/codecs/rt5640.c
View File

@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
static const struct regmap_config rt5640_regmap = {
.reg_bits = 8,
.val_bits = 16,
.use_single_rw = true,
.max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
RT5640_PR_SPACING),

8
sound/soc/codecs/rt5677.c
View File

@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "BST2", NULL, "IN2P" },
{ "BST2", NULL, "IN2N" },
{ "IN1P", NULL, "micbias1" },
{ "IN1N", NULL, "micbias1" },
{ "IN2P", NULL, "micbias1" },
{ "IN2N", NULL, "micbias1" },
{ "IN1P", NULL, "MICBIAS1" },
{ "IN1N", NULL, "MICBIAS1" },
{ "IN2P", NULL, "MICBIAS1" },
{ "IN2N", NULL, "MICBIAS1" },
{ "ADC 1", NULL, "BST1" },
{ "ADC 1", NULL, "ADC 1 power" },

8
sound/soc/generic/simple-card.c
View File

@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->snd_card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
if (ret >= 0)
return ret;
err:
asoc_simple_card_unref(pdev);
return ret;
}
static int asoc_simple_card_remove(struct platform_device *pdev)
{
return asoc_simple_card_unref(pdev);
}
static const struct of_device_id asoc_simple_of_match[] = {
{ .compatible = "simple-audio-card", },
{},
@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
.remove = asoc_simple_card_remove,
};
module_platform_driver(asoc_simple_card);

2
sound/soc/omap/omap-twl4030.c
View File

@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
.platform_name = "omap-mcbsp.2",
.platform_name = "omap-mcbsp.3",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,

2
sound/soc/sh/rcar/gen.c
View File

@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
};
/* it shouldn't happen */
if (use_dvc & !use_src)
if (use_dvc && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */

2
sound/soc/soc-core.c
View File

@ -1325,7 +1325,7 @@ 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->init_name = name;
dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);

2
sound/soc/tegra/tegra_asoc_utils.h
View File

@ -21,7 +21,7 @@
*/
#ifndef __TEGRA_ASOC_UTILS_H__
#define __TEGRA_ASOC_UTILS_H_
#define __TEGRA_ASOC_UTILS_H__
struct clk;
struct device;