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author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-10-02 16:36:10 (GMT) |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-10-02 16:36:10 (GMT) |
commit | be0f52baa4700ff73dfc528dc8d56fad1995bd02 (patch) | |
tree | 785fc31d4518f040e8b7066d85ee4ed8f68bbb54 | |
parent | 6acf9902a21097a8989a2343bc3bbe14cb2322ce (diff) | |
parent | 4a4370442c996be0fd08234a167c8a127c2488bb (diff) | |
download | linux-fsl-qoriq-be0f52baa4700ff73dfc528dc8d56fad1995bd02.tar.xz |
Merge tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All small, mostly driver-specific fixes: a few ASoC driver fixes
(trivial stable fixes, sgtl5000 fixes), one DPCM fix, an old AC97 ID,
and a fix for HD-audio Conexant GPIO"
* tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix GPIO for Acer Aspire 3830TG
ALSA: ac97: Add ID for TI TLV320AIC27 codec
ASoC: imx-sgtl5000: Fix uninitialized pointer use in error path
ASoC: imx-sgtl5000: do not use devres on a foreign device
ASoC: blackfin: Add missing break statement to bf6xx
ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st()
ASoC: ab8500-codec: info leak in anc_status_control_put()
ASoC: max98095: a couple array underflows
ASoC: core: Only add platform DAI widgets once.
-rw-r--r-- | sound/pci/ac97/ac97_codec.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 11 | ||||
-rw-r--r-- | sound/soc/blackfin/bf6xx-i2s.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/88pm860x-codec.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/ab8500-codec.c | 7 | ||||
-rw-r--r-- | sound/soc/codecs/max98095.c | 4 | ||||
-rw-r--r-- | sound/soc/fsl/imx-sgtl5000.c | 7 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 1 |
8 files changed, 29 insertions, 6 deletions
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4edd2d0..ec68eae 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,7 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { 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(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c index c02405c..5810a06 100644 --- a/sound/soc/blackfin/bf6xx-i2s.c +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -88,6 +88,7 @@ static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S8: param.spctl |= 0x70; sport->wdsize = 1; + break; case SNDRV_PCM_FORMAT_S16_LE: param.spctl |= 0xf0; sport->wdsize = 2; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af0434..259d1ac 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err; diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0ad..80555d7 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd16..8dbcacd 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..ca1be1d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data; + struct imx_sgtl5000_data *data = NULL; int int_port, ext_port; int ret; @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (data && !IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); snd_soc_unregister_card(&data->card); + clk_put(data->codec_clk); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d05613..1a38be0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1380,7 +1380,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return -ENODEV; list_add(&cpu_dai->dapm.list, &card->dapm_list); - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } if (cpu_dai->driver->probe) { |