From: Takashi Iwai Date: Mon, 27 Oct 2008 16:08:11 +0000 (+0100) Subject: Merge branches 'topic/fix/asoc', 'topic/fix/hda', 'topic/fix/misc' and 'topic/pci... X-Git-Tag: v2.6.28-rc3~97^2 X-Git-Url: http://www.pilppa.org/gitweb/gitweb.cgi?a=commitdiff_plain;h=0a9b86381c76e9d33a9b6edb66aef32d7a3306e3;hp=2f5ad54ea6e2e38156bfb889964deee991f3087a;p=linux-2.6-omap-h63xx.git Merge branches 'topic/fix/asoc', 'topic/fix/hda', 'topic/fix/misc' and 'topic/pci-ioremap-bar' into for-linus --- diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index e6beb92c693..b4590df0746 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; } diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb77cb9..75a0d746fb6 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index eb9bc365530..c180598f171 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -1,7 +1,7 @@ /* * Initialisation code for Cyrix/NatSemi VSA1 softaudio * - * (C) Copyright 2003 Red Hat Inc + * (C) Copyright 2003 Red Hat Inc * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 4d9378d8120..6dea5b5cc77 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -2,7 +2,7 @@ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio * * (C) Copyright 2007 Ash Willis - * (C) Copyright 2003 Red Hat Inc + * (C) Copyright 2003 Red Hat Inc * * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did * mess with it a bit. The chip seems to have to have trouble with full duplex diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e72707cb60a..4eceab9bd10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -307,6 +307,13 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; + +#ifdef SND_HDA_NEEDS_RESUME +#define ALC_MAX_PINS 16 + unsigned int num_pins; + hda_nid_t pin_nids[ALC_MAX_PINS]; + unsigned int pin_cfgs[ALC_MAX_PINS]; +#endif }; /* @@ -2778,6 +2785,64 @@ static void alc_free(struct hda_codec *codec) codec->spec = NULL; /* to be sure */ } +#ifdef SND_HDA_NEEDS_RESUME +static void store_pin_configs(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid, end_nid; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = + (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wid_type != AC_WID_PIN) + continue; + if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) + break; + spec->pin_nids[spec->num_pins] = nid; + spec->pin_cfgs[spec->num_pins] = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + spec->num_pins++; + } +} + +static void resume_pin_configs(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) { + hda_nid_t pin_nid = spec->pin_nids[i]; + unsigned int pin_config = spec->pin_cfgs[i]; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + } +} + +static int alc_resume(struct hda_codec *codec) +{ + resume_pin_configs(codec); + codec->patch_ops.init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + return 0; +} +#else +#define store_pin_configs(codec) +#endif + /* */ static struct hda_codec_ops alc_patch_ops = { @@ -2786,6 +2851,9 @@ static struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = alc_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = alc_resume, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = alc_check_power_status, #endif @@ -3832,6 +3900,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + store_pin_configs(codec); return 1; } @@ -4996,7 +5065,7 @@ static struct hda_verb alc260_test_init_verbs[] = { */ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, - const char *pfx) + const char *pfx, int *vol_bits) { hda_nid_t nid_vol; unsigned long vol_val, sw_val; @@ -5018,10 +5087,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, } else return 0; /* N/A */ - snprintf(name, sizeof(name), "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); - if (err < 0) - return err; + if (!(*vol_bits & (1 << nid_vol))) { + /* first control for the volume widget */ + snprintf(name, sizeof(name), "%s Playback Volume", pfx); + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); + if (err < 0) + return err; + *vol_bits |= (1 << nid_vol); + } snprintf(name, sizeof(name), "%s Playback Switch", pfx); err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val); if (err < 0) @@ -5035,6 +5108,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, { hda_nid_t nid; int err; + int vols = 0; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = spec->private_dac_nids; @@ -5042,21 +5116,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Front"); + err = alc260_add_playback_controls(spec, nid, "Front", &vols); if (err < 0) return err; } nid = cfg->speaker_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Speaker"); + err = alc260_add_playback_controls(spec, nid, "Speaker", &vols); if (err < 0) return err; } nid = cfg->hp_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Headphone"); + err = alc260_add_playback_controls(spec, nid, "Headphone", + &vols); if (err < 0) return err; } @@ -5244,6 +5319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) } spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -10307,6 +10383,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -11441,6 +11518,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -12224,6 +12302,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc269_capture_mixer; spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -13310,6 +13389,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc861_capture_mixer; spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -14421,6 +14501,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -16252,6 +16333,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; + + store_pin_configs(codec); return 1; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2ac7205d45..788fdc6f326 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1282,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; spec->multiout.share_spdif = 1; } - if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) { + if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); if (err < 0) return err; diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 827587f0818..e020c160ee4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = { } }; -static u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, -}; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) @@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 05336ed7e49..cff276ee261 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33ae343..8485a8a9d0f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. diff --git a/sound/sound_core.c b/sound/sound_core.c index faef87a9bc3..a75b289a5d7 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -57,7 +57,7 @@ module_exit(cleanup_soundcore); /* * OSS sound core handling. Breaks out sound functions to submodules * - * Author: Alan Cox + * Author: Alan Cox * * Fixes: *