mirror of https://gitee.com/openkylin/linux.git
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: Fix Trivial Warnining in sound/pci/cmipci.c ALSA: snd-usb-caiaq: fix reported elapsed periods ASoC: s3c-i2s-v2 needs to declare a license for modular builds ALSA: hda - Fix init verbs of AD1884A mobile model ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991 ASoC: Fix WM8580 volume update handling for large register changes ASoC: Fix offset of freqmode in WM8580 PLL configuration
This commit is contained in:
commit
14b6084daa
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@ -3014,7 +3014,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
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.dev_free = snd_cmipci_dev_free,
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};
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unsigned int val;
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long iomidi;
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long iomidi = 0;
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int integrated_midi = 0;
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char modelstr[16];
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int pcm_index, pcm_spdif_index;
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@ -3817,6 +3817,49 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
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{ } /* end */
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};
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static struct hda_verb ad1884a_mobile_verbs[] = {
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/* DACs; unmute as default */
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{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
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{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
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/* Port-A (HP) mixer - route only from analog mixer */
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{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
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{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
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/* Port-A pin */
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{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
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/* Port-A (HP) pin - always unmuted */
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{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
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/* Port-B (mic jack) pin */
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{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
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{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
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/* Port-C (int mic) pin */
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{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
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{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
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/* Port-F (int speaker) mixer - route only from analog mixer */
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{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
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{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
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/* Port-F pin */
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{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
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{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
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/* Analog mixer; mute as default */
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
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{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
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/* Analog Mix output amp */
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{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
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/* capture sources */
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/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
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{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
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{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
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{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
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/* unsolicited event for pin-sense */
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{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
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{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
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{ } /* end */
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};
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/*
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* Thinkpad X300
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* 0x11 - HP
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@ -3988,7 +4031,7 @@ static int patch_ad1884a(struct hda_codec *codec)
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break;
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case AD1884A_MOBILE:
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spec->mixers[0] = ad1884a_mobile_mixers;
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spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
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spec->init_verbs[0] = ad1884a_mobile_verbs;
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spec->multiout.dig_out_nid = 0;
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codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
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codec->patch_ops.init = ad1884a_hp_init;
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@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
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obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
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obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
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obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
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obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o
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obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
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obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
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obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
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@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
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static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct soc_mixer_control *mc =
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(struct soc_mixer_control *)kcontrol->private_value;
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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int reg = kcontrol->private_value & 0xff;
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int reg2 = (kcontrol->private_value >> 24) & 0xff;
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unsigned int reg = mc->reg;
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unsigned int reg2 = mc->rreg;
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int ret;
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u16 val;
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@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
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return 0;
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}
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#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
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#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
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xinvert, tlv_array) \
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{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
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.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
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SNDRV_CTL_ELEM_ACCESS_READWRITE, \
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.tlv.p = (tlv_array), \
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.info = snd_soc_info_volsw_2r, \
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.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
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.private_value = (reg_left) | ((shift) << 8) | \
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((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
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.private_value = (unsigned long)&(struct soc_mixer_control) \
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{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
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.max = xmax, .invert = xinvert} }
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static const struct snd_kcontrol_new wm8580_snd_controls[] = {
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SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
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@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
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reg = wm8580_read(codec, WM8580_PLLA4 + offset);
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reg &= ~0x3f;
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reg |= pll_div.prescale | pll_div.postscale << 1 |
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pll_div.freqmode << 4;
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pll_div.freqmode << 3;
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wm8580_write(codec, WM8580_PLLA4 + offset, reg);
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@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
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return snd_soc_register_dai(dai);
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}
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EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
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MODULE_LICENSE("GPL");
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@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
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debug("%s(%p)\n", __func__, substream);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
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dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
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else
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} else {
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dev->period_in_count[index] = BYTES_PER_SAMPLE;
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dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
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}
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if (dev->streaming)
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return 0;
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@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
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if (!sub)
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continue;
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pb = frames_to_bytes(sub->runtime,
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sub->runtime->period_size);
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pb = snd_pcm_lib_period_bytes(sub);
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cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
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&dev->period_out_count[stream] :
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&dev->period_in_count[stream];
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@ -35,7 +35,7 @@
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#include "input.h"
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MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
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MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13");
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MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14");
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MODULE_LICENSE("GPL");
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MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
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"{Native Instruments, RigKontrol3},"
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