Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Don't query connections for widgets have no connections
  ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
  ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
  ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
  ALSA: HDA: Fix dock mic for Lenovo X220-tablet
  ASoC: format_register_str: Don't clip register values
  ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
  ASoC: zylonite: set .codec_dai_name in initializer

diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 8808b82..76e0d56 100644 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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;
                /*