diff options
author | Takashi Iwai <tiwai@suse.de> | 2012-02-16 16:43:09 +0100 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2012-02-16 16:43:09 +0100 |
commit | 00bc0ce9130551ef193c3f5db0b7b6e70dff28ac (patch) | |
tree | b6b022250a08073e522ed1bba56586b534a3c77e | |
parent | a7f3eedc88b547e0ec35ba4cc4ae61cd9bc760ac (diff) | |
parent | c14c95f62ecb8710af14ae0d48e01991b70bb6f4 (diff) |
Merge branch 'fix/hda' into topic/hda
The fix for bitmap-overflow in Realtek codec driver is needed for the
further development of the auto-parser with badness evaluation.
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 34 | ||||
-rw-r--r-- | sound/pci/hda/patch_via.c | 3 | ||||
-rw-r--r-- | sound/pci/intel8x0.c | 6 | ||||
-rw-r--r-- | sound/soc/sh/fsi.c | 6 | ||||
-rw-r--r-- | sound/usb/card.h | 1 | ||||
-rw-r--r-- | sound/usb/format.c | 4 | ||||
-rw-r--r-- | sound/usb/quirks.c | 6 |
7 files changed, 48 insertions, 12 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b8e06eb96e1..0ffccc17895 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -80,6 +80,8 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; +#define MAX_VOL_NIDS 0x40 + struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -118,8 +120,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ - DECLARE_BITMAP(vol_ctls, 0x20 << 1); - DECLARE_BITMAP(sw_ctls, 0x20 << 1); + DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -3125,7 +3127,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) static inline unsigned int get_ctl_pos(unsigned int data) { hda_nid_t nid = get_amp_nid_(data); - unsigned int dir = get_amp_direction_(data); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); return (nid << 1) | dir; } @@ -4399,6 +4404,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_8930G, ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, + ALC889_FIXUP_DAC_ROUTE, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -4452,6 +4458,23 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec, alc882_gpio_mute(codec, 1, 0); } +/* Fix the connection of some pins for ALC889: + * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't + * work correctly (bko#42740) + */ +static void alc889_fixup_dac_route(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn1[2] = { 0x0c, 0x0d }; + hda_nid_t conn2[2] = { 0x0e, 0x0f }; + snd_hda_override_conn_list(codec, 0x14, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4599,6 +4622,10 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc885_fixup_macpro_gpio, }, + [ALC889_FIXUP_DAC_ROUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_dac_route, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4623,6 +4650,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e5842fe1b1e..c7eb4d7d05c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init input-src */ for (i = 0; i < spec->num_adc_nids; i++) { int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx; + /* secondary ADCs must have the unique MUX */ + if (i > 0 && !spec->mux_nids[i]) + break; if (spec->mux_nids[adc_idx]) { int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx; snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9f3b01bb72c..e0a4263baa2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x161f, + .subdevice = 0x202f, + .name = "Gateway M520", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x161f, .subdevice = 0x203a, .name = "Gateway 4525GZ", /* AD1981B */ .type = AC97_TUNE_INV_EAPD diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a28bd..ea4a82d0116 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); - int samples_pos = io->buff_sample_pos - 1; - if (samples_pos < 0) - samples_pos = 0; - - return fsi_sample2frame(fsi, samples_pos); + return fsi_sample2frame(fsi, io->buff_sample_pos); } static struct snd_pcm_ops fsi_pcm_ops = { diff --git a/sound/usb/card.h b/sound/usb/card.h index a39edcc32a9..da5fa1ac4ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -1,6 +1,7 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H +#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/sound/usb/format.c b/sound/usb/format.c index e09aba19375..ddfef57c4c9 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } -#define MAX_UAC2_NR_RATES 1024 - /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_UAC2_NR_RATES) { + if (nr_rates >= MAX_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a3ddac0deff..27817266867 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { + if (!fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } + if (fp->nr_rates > MAX_NR_RATES) { + kfree(fp); + return -EINVAL; + } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); |