[GIT PULL] sound fixes for 2.6.39-rc3

From: Takashi Iwai
Date: Sun Apr 10 2011 - 05:59:33 EST

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Linus,

please pull ALSA updates for v2.6.39-rc3 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

All small fixes as belw.

Thanks!

Takashi

===

Aaron Plattner (1):
ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums

Antonio Ospite (1):
ASoC: zylonite: set .codec_dai_name in initializer

David Henningsson (2):
ALSA: HDA: Fix dock mic for Lenovo X220-tablet
ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)

Stephen Warren (1):
ASoC: format_register_str: Don't clip register values

Takashi Iwai (1):
ALSA: hda - Don't query connections for widgets have no connections

Tarek Soliman (1):
ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable

Vasily Khoruzhick (1):
ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare

---
sound/arm/pxa2xx-pcm-lib.c | 3 ++
sound/pci/hda/patch_conexant.c | 1 +
sound/pci/hda/patch_hdmi.c | 70 +++++++++++++++++++++++++---------------
sound/pci/hda/patch_realtek.c | 2 +-
sound/pci/hda/patch_sigmatel.c | 3 ++
sound/soc/pxa/pxa2xx-pcm.c | 1 +
sound/soc/pxa/zylonite.c | 6 ++--
sound/soc/soc-core.c | 8 ++--
sound/usb/midi.c | 1 +
9 files changed, 61 insertions(+), 34 deletions(-)

diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 8808b82..76e0d56 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -140,6 +140,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
if (!prtd || !prtd->params)
return 0;

+ if (prtd->dma_ch == -1)
+ return -EINVAL;
+
DCSR(prtd->dma_ch) &= ~DCSR_RUN;
DCSR(prtd->dma_ch) = 0;
DCMD(prtd->dma_ch) = 0;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 69e3386..ad97d93 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3035,6 +3035,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
{}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 251773e..715615a 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
stream_tag, format, substream);
}

+static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec,
+ int channels)
+{
+ unsigned int chanmask;
+ int chan = channels ? (channels - 1) : 1;
+
+ switch (channels) {
+ default:
+ case 0:
+ case 2:
+ chanmask = 0x00;
+ break;
+ case 4:
+ chanmask = 0x08;
+ break;
+ case 6:
+ chanmask = 0x0b;
+ break;
+ case 8:
+ chanmask = 0x13;
+ break;
+ }
+
+ /* Set the audio infoframe channel allocation and checksum fields. The
+ * channel count is computed implicitly by the hardware. */
+ snd_hda_codec_write(codec, 0x1, 0,
+ Nv_VERB_SET_Channel_Allocation, chanmask);
+
+ snd_hda_codec_write(codec, 0x1, 0,
+ Nv_VERB_SET_Info_Frame_Checksum,
+ (0x71 - chan - chanmask));
+}
+
static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
@@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
AC_VERB_SET_STREAM_FORMAT, 0);
}

+ /* The audio hardware sends a channel count of 0x7 (8ch) when all the
+ * streams are disabled. */
+ nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}

@@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
int chs;
- unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id;
+ unsigned int dataDCC1, dataDCC2, channel_id;
int i;

mutex_lock(&codec->spdif_mutex);

chs = substream->runtime->channels;
- chan = chs ? (chs - 1) : 1;

- switch (chs) {
- default:
- case 0:
- case 2:
- chanmask = 0x00;
- break;
- case 4:
- chanmask = 0x08;
- break;
- case 6:
- chanmask = 0x0b;
- break;
- case 8:
- chanmask = 0x13;
- break;
- }
dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
dataDCC2 = 0x2;

- /* set the Audio InforFrame Channel Allocation */
- snd_hda_codec_write(codec, 0x1, 0,
- Nv_VERB_SET_Channel_Allocation, chanmask);
-
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
snd_hda_codec_write(codec,
@@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
}
}

- /* set the Audio Info Frame Checksum */
- snd_hda_codec_write(codec, 0x1, 0,
- Nv_VERB_SET_Info_Frame_Checksum,
- (0x71 - chan - chanmask));
+ nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs);

mutex_unlock(&codec->spdif_mutex);
return 0;
@@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
spec->multiout.max_channels = 8;
spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x;
codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
+
+ /* Initialize the audio infoframe channel mask and checksum to something
+ * valid */
+ nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
return 0;
}

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7e28a64..52928d9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14124,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
};

static hda_nid_t alc269_adc_candidates[] = {
- 0x08, 0x09, 0x07,
+ 0x08, 0x09, 0x07, 0x11,
};

#define alc269_modes alc260_modes
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1395991..94d19c0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3408,6 +3408,9 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;

+ if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
+ return -1;
+
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 02fb664..2ce0b2d 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
if (prtd->dma_ch >= 0) {
pxa_free_dma(prtd->dma_ch);
prtd->dma_ch = -1;
+ prtd->params = NULL;
}

return 0;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ac57726..b644575 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa2xx-ac97",
- .codec_name = "wm9713-hifi",
+ .codec_dai_name = "wm9713-hifi",
.init = zylonite_wm9713_init,
},
{
@@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa2xx-ac97-aux",
- .codec_name = "wm9713-aux",
+ .codec_dai_name = "wm9713-aux",
},
{
.name = "WM9713 Voice",
@@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
.cpu_dai_name = "pxa-ssp-dai.2",
- .codec_name = "wm9713-voice",
+ .codec_dai_name = "wm9713-voice",
.ops = &zylonite_voice_ops,
},
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4dda589..b76b74d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val)
static int format_register_str(struct snd_soc_codec *codec,
unsigned int reg, char *buf, size_t len)
{
- int wordsize = codec->driver->reg_word_size * 2;
- int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ int regsize = codec->driver->reg_word_size * 2;
int ret;
char tmpbuf[len + 1];
char regbuf[regsize + 1];
@@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
size_t total = 0;
loff_t p = 0;

- wordsize = codec->driver->reg_word_size * 2;
- regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ regsize = codec->driver->reg_word_size * 2;

len = wordsize + regsize + 2 + 1;

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b4b39c0..f928910 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1301,6 +1301,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */
case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */
case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */
+ case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */
ep->max_transfer = 4;
break;
/*
--
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Next message: Takashi Iwai: "Re: linux-next: manual merge of the sound tree with Linus' tree"
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Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]