Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits) ALSA: usb - Use strlcat() correctly ALSA: Fix invalid __exit in sound/mips/*.c ALSA: hda - Fix / improve ALC66x parser ALSA: ctxfi: Swapped SURROUND-SIDE mute sound: Make keywest_driver static ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs ASoC: fix kconfig order of Blackfin drivers ALSA: hda - Added quirk to enable sound on Toshiba NB200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 ALSA: Don't assume i2c device probing always succeeds ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P ALSA: echoaudio - Re-enable the line-out control for the Mia card ALSA: hda - Resurrect input-source mixer of ALC268 model=acer ALSA: hda - Analog Devices AD1984A add HP Touchsmart model ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist ALSA: hda - CD-audio sound for hda-intel conexant benq laptop ASoC: DaVinci: Correct McASP FIFO initialization ASoC: Davinci: Fix race with cpu_dai->dma_data ASoC: DaVinci: Fix divide by zero error during 1st execution ...
This commit is contained in:
commit
f0a221ef47
|
@ -209,6 +209,7 @@ AD1884A / AD1883 / AD1984A / AD1984B
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laptop laptop with HP jack sensing
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mobile mobile devices with HP jack sensing
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thinkpad Lenovo Thinkpad X300
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touchsmart HP Touchsmart
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AD1884
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======
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|
|
|
@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter,
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client = i2c_new_device(adapter, &info);
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if (!client)
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return -ENODEV;
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/*
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* We know the driver is already loaded, so the device should be
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* already bound. If not it means binding failed, and then there
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* is no point in keeping the device instantiated.
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*/
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if (!client->driver) {
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i2c_unregister_device(client);
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return -ENODEV;
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}
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/*
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* Let i2c-core delete that device on driver removal.
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|
|
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@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev)
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return 0;
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}
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static int __exit hal2_remove(struct platform_device *pdev)
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static int __devexit hal2_remove(struct platform_device *pdev)
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{
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struct snd_card *card = platform_get_drvdata(pdev);
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@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
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return 0;
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}
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static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
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static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
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{
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struct snd_card *card = platform_get_drvdata(pdev);
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|
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@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
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static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
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{
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return atc_daio_unmute(atc, state, LINEO4);
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return atc_daio_unmute(atc, state, LINEO2);
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}
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static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
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@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
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static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
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{
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return atc_daio_unmute(atc, state, LINEO2);
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return atc_daio_unmute(atc, state, LINEO4);
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}
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static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
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|
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@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
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Control interface
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******************************************************************************/
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#ifndef ECHOCARD_HAS_VMIXER
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#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
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/******************* PCM output volume *******************/
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static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
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@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
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return changed;
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}
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#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
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/* On the Mia this one controls the line-out volume */
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static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
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.name = "Line Playback Volume",
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
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SNDRV_CTL_ELEM_ACCESS_TLV_READ,
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.info = snd_echo_output_gain_info,
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.get = snd_echo_output_gain_get,
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.put = snd_echo_output_gain_put,
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.tlv = {.p = db_scale_output_gain},
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};
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#else
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static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
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.name = "PCM Playback Volume",
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
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|
@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
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.put = snd_echo_output_gain_put,
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.tlv = {.p = db_scale_output_gain},
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};
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#endif
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#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
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#ifdef ECHOCARD_HAS_INPUT_GAIN
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|
@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
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snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
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if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
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goto ctl_error;
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#else
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if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
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#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&snd_echo_line_output_gain, chip));
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if (err < 0)
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goto ctl_error;
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#endif
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#else /* ECHOCARD_HAS_VMIXER */
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err = snd_ctl_add(chip->card,
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snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
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if (err < 0)
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goto ctl_error;
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#endif /* ECHOCARD_HAS_VMIXER */
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#ifdef ECHOCARD_HAS_INPUT_GAIN
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if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
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|
|
|
@ -29,6 +29,7 @@
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#define ECHOCARD_HAS_ADAT FALSE
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#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
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#define ECHOCARD_HAS_MIDI
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#define ECHOCARD_HAS_LINE_OUT_GAIN
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/* Pipe indexes */
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#define PX_ANALOG_OUT 0 /* 8 */
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|
|
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@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
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* white-list for enable_msi
|
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*/
|
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static struct snd_pci_quirk msi_white_list[] __devinitdata = {
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SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1),
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SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
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{}
|
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};
|
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|
|
|
@ -4031,6 +4031,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
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return 0;
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}
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|
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/*
|
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* HP Touchsmart
|
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* port-A (0x11) - front hp-out
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* port-B (0x14) - unused
|
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* port-C (0x15) - unused
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* port-D (0x12) - rear line out
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* port-E (0x1c) - front mic-in
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* port-F (0x16) - Internal speakers
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* digital-mic (0x17) - Internal mic
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*/
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static struct hda_verb ad1984a_touchsmart_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-E (int speaker) mixer - route only from analog mixer */
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{0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
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/* Port-E pin */
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{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
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{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
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{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
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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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{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
|
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/* allow to touch GPIO1 (for mute control) */
|
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{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
|
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{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
|
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{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
|
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/* internal mic - dmic */
|
||||
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
|
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/* set magic COEFs for dmic */
|
||||
{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
|
||||
{0x01, AC_VERB_SET_PROC_COEF, 0x08},
|
||||
{ } /* end */
|
||||
};
|
||||
|
||||
static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
|
||||
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
|
||||
/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
|
||||
{
|
||||
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
|
||||
.name = "Master Playback Switch",
|
||||
.info = snd_hda_mixer_amp_switch_info,
|
||||
.get = snd_hda_mixer_amp_switch_get,
|
||||
.put = ad1884a_mobile_master_sw_put,
|
||||
.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
|
||||
},
|
||||
HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
|
||||
HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
|
||||
HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
|
||||
HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
|
||||
HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
|
||||
{ } /* end */
|
||||
};
|
||||
|
||||
/* switch to external mic if plugged */
|
||||
static void ad1984a_touchsmart_automic(struct hda_codec *codec)
|
||||
{
|
||||
if (snd_hda_codec_read(codec, 0x1c, 0,
|
||||
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) {
|
||||
snd_hda_codec_write(codec, 0x0c, 0,
|
||||
AC_VERB_SET_CONNECT_SEL, 0x4);
|
||||
} else {
|
||||
snd_hda_codec_write(codec, 0x0c, 0,
|
||||
AC_VERB_SET_CONNECT_SEL, 0x5);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
/* unsolicited event for HP jack sensing */
|
||||
static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
|
||||
unsigned int res)
|
||||
{
|
||||
switch (res >> 26) {
|
||||
case AD1884A_HP_EVENT:
|
||||
ad1884a_hp_automute(codec);
|
||||
break;
|
||||
case AD1884A_MIC_EVENT:
|
||||
ad1984a_touchsmart_automic(codec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* initialize jack-sensing, too */
|
||||
static int ad1984a_touchsmart_init(struct hda_codec *codec)
|
||||
{
|
||||
ad198x_init(codec);
|
||||
ad1884a_hp_automute(codec);
|
||||
ad1984a_touchsmart_automic(codec);
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
/*
|
||||
*/
|
||||
|
||||
|
@ -4039,6 +4160,7 @@ enum {
|
|||
AD1884A_LAPTOP,
|
||||
AD1884A_MOBILE,
|
||||
AD1884A_THINKPAD,
|
||||
AD1984A_TOUCHSMART,
|
||||
AD1884A_MODELS
|
||||
};
|
||||
|
||||
|
@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
|
|||
[AD1884A_LAPTOP] = "laptop",
|
||||
[AD1884A_MOBILE] = "mobile",
|
||||
[AD1884A_THINKPAD] = "thinkpad",
|
||||
[AD1984A_TOUCHSMART] = "touchsmart",
|
||||
};
|
||||
|
||||
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
|
||||
|
@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
|
|||
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
|
||||
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
|
||||
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
|
||||
SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
|
||||
{}
|
||||
};
|
||||
|
||||
|
@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec)
|
|||
codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
|
||||
codec->patch_ops.init = ad1984a_thinkpad_init;
|
||||
break;
|
||||
case AD1984A_TOUCHSMART:
|
||||
spec->mixers[0] = ad1984a_touchsmart_mixers;
|
||||
spec->init_verbs[0] = ad1984a_touchsmart_verbs;
|
||||
spec->multiout.dig_out_nid = 0;
|
||||
codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
|
||||
codec->patch_ops.init = ad1984a_touchsmart_init;
|
||||
/* set the upper-limit for mixer amp to 0dB for avoiding the
|
||||
* possible damage by overloading
|
||||
*/
|
||||
snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
|
||||
(0x17 << AC_AMPCAP_OFFSET_SHIFT) |
|
||||
(0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
|
||||
(0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
|
||||
(1 << AC_AMPCAP_MUTE_SHIFT));
|
||||
break;
|
||||
}
|
||||
|
||||
return 0;
|
||||
|
|
|
@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = {
|
|||
};
|
||||
|
||||
static struct hda_input_mux cxt5045_capture_source_benq = {
|
||||
.num_items = 3,
|
||||
.num_items = 5,
|
||||
.items = {
|
||||
{ "IntMic", 0x1 },
|
||||
{ "ExtMic", 0x2 },
|
||||
{ "LineIn", 0x3 },
|
||||
{ "CD", 0x4 },
|
||||
{ "Mixer", 0x0 },
|
||||
}
|
||||
};
|
||||
|
||||
|
@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
|
|||
};
|
||||
|
||||
static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
|
||||
HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
|
||||
HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
|
||||
|
||||
HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
|
||||
HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
|
||||
|
||||
HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
|
||||
HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
|
||||
|
||||
{}
|
||||
};
|
||||
|
||||
|
|
|
@ -12660,7 +12660,7 @@ static struct alc_config_preset alc268_presets[] = {
|
|||
.init_hook = alc268_toshiba_automute,
|
||||
},
|
||||
[ALC268_ACER] = {
|
||||
.mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
|
||||
.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
|
||||
alc268_beep_mixer },
|
||||
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
|
||||
alc268_acer_verbs },
|
||||
|
@ -16852,6 +16852,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
|
|||
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
|
||||
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
|
||||
ALC662_3ST_6ch_DIG),
|
||||
SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
|
||||
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
|
||||
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
|
||||
ALC662_3ST_6ch_DIG),
|
||||
|
@ -17145,70 +17146,145 @@ static struct alc_config_preset alc662_presets[] = {
|
|||
* BIOS auto configuration
|
||||
*/
|
||||
|
||||
/* add playback controls from the parsed DAC table */
|
||||
static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
|
||||
/* convert from MIX nid to DAC */
|
||||
static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
|
||||
{
|
||||
if (nid == 0x0f)
|
||||
return 0x02;
|
||||
else if (nid >= 0x0c && nid <= 0x0e)
|
||||
return nid - 0x0c + 0x02;
|
||||
else
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* get MIX nid connected to the given pin targeted to DAC */
|
||||
static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
|
||||
hda_nid_t dac)
|
||||
{
|
||||
hda_nid_t mix[4];
|
||||
int i, num;
|
||||
|
||||
num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
|
||||
for (i = 0; i < num; i++) {
|
||||
if (alc662_mix_to_dac(mix[i]) == dac)
|
||||
return mix[i];
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* look for an empty DAC slot */
|
||||
static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
|
||||
{
|
||||
struct alc_spec *spec = codec->spec;
|
||||
hda_nid_t srcs[5];
|
||||
int i, j, num;
|
||||
|
||||
num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
|
||||
if (num < 0)
|
||||
return 0;
|
||||
for (i = 0; i < num; i++) {
|
||||
hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
|
||||
if (!nid)
|
||||
continue;
|
||||
for (j = 0; j < spec->multiout.num_dacs; j++)
|
||||
if (spec->multiout.dac_nids[j] == nid)
|
||||
break;
|
||||
if (j >= spec->multiout.num_dacs)
|
||||
return nid;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* fill in the dac_nids table from the parsed pin configuration */
|
||||
static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
|
||||
const struct auto_pin_cfg *cfg)
|
||||
{
|
||||
struct alc_spec *spec = codec->spec;
|
||||
int i;
|
||||
hda_nid_t dac;
|
||||
|
||||
spec->multiout.dac_nids = spec->private_dac_nids;
|
||||
for (i = 0; i < cfg->line_outs; i++) {
|
||||
dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
|
||||
if (!dac)
|
||||
continue;
|
||||
spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
|
||||
hda_nid_t nid, unsigned int chs)
|
||||
{
|
||||
char name[32];
|
||||
sprintf(name, "%s Playback Volume", pfx);
|
||||
return add_control(spec, ALC_CTL_WIDGET_VOL, name,
|
||||
HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
|
||||
}
|
||||
|
||||
static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
|
||||
hda_nid_t nid, unsigned int chs)
|
||||
{
|
||||
char name[32];
|
||||
sprintf(name, "%s Playback Switch", pfx);
|
||||
return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
|
||||
HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
|
||||
}
|
||||
|
||||
#define alc662_add_stereo_vol(spec, pfx, nid) \
|
||||
alc662_add_vol_ctl(spec, pfx, nid, 3)
|
||||
#define alc662_add_stereo_sw(spec, pfx, nid) \
|
||||
alc662_add_sw_ctl(spec, pfx, nid, 3)
|
||||
|
||||
/* add playback controls from the parsed DAC table */
|
||||
static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
|
||||
const struct auto_pin_cfg *cfg)
|
||||
{
|
||||
struct alc_spec *spec = codec->spec;
|
||||
static const char *chname[4] = {
|
||||
"Front", "Surround", NULL /*CLFE*/, "Side"
|
||||
};
|
||||
hda_nid_t nid;
|
||||
hda_nid_t nid, mix;
|
||||
int i, err;
|
||||
|
||||
for (i = 0; i < cfg->line_outs; i++) {
|
||||
if (!spec->multiout.dac_nids[i])
|
||||
nid = spec->multiout.dac_nids[i];
|
||||
if (!nid)
|
||||
continue;
|
||||
mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
|
||||
if (!mix)
|
||||
continue;
|
||||
nid = alc880_idx_to_dac(i);
|
||||
if (i == 2) {
|
||||
/* Center/LFE */
|
||||
err = add_control(spec, ALC_CTL_WIDGET_VOL,
|
||||
"Center Playback Volume",
|
||||
HDA_COMPOSE_AMP_VAL(nid, 1, 0,
|
||||
HDA_OUTPUT));
|
||||
err = alc662_add_vol_ctl(spec, "Center", nid, 1);
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = add_control(spec, ALC_CTL_WIDGET_VOL,
|
||||
"LFE Playback Volume",
|
||||
HDA_COMPOSE_AMP_VAL(nid, 2, 0,
|
||||
HDA_OUTPUT));
|
||||
err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
|
||||
"Center Playback Switch",
|
||||
HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
|
||||
HDA_INPUT));
|
||||
err = alc662_add_sw_ctl(spec, "Center", mix, 1);
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
|
||||
"LFE Playback Switch",
|
||||
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
|
||||
HDA_INPUT));
|
||||
err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
|
||||
if (err < 0)
|
||||
return err;
|
||||
} else {
|
||||
const char *pfx;
|
||||
if (cfg->line_outs == 1 &&
|
||||
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
|
||||
if (!cfg->hp_pins)
|
||||
if (cfg->hp_outs)
|
||||
pfx = "Speaker";
|
||||
else
|
||||
pfx = "PCM";
|
||||
} else
|
||||
pfx = chname[i];
|
||||
sprintf(name, "%s Playback Volume", pfx);
|
||||
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
|
||||
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
|
||||
HDA_OUTPUT));
|
||||
err = alc662_add_vol_ctl(spec, pfx, nid, 3);
|
||||
if (err < 0)
|
||||
return err;
|
||||
if (cfg->line_outs == 1 &&
|
||||
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
|
||||
pfx = "Speaker";
|
||||
sprintf(name, "%s Playback Switch", pfx);
|
||||
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
|
||||
HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
|
||||
3, 0, HDA_INPUT));
|
||||
err = alc662_add_sw_ctl(spec, pfx, mix, 3);
|
||||
if (err < 0)
|
||||
return err;
|
||||
}
|
||||
|
@ -17217,54 +17293,38 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
|
|||
}
|
||||
|
||||
/* add playback controls for speaker and HP outputs */
|
||||
static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
|
||||
/* return DAC nid if any new DAC is assigned */
|
||||
static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
|
||||
const char *pfx)
|
||||
{
|
||||
hda_nid_t nid;
|
||||
struct alc_spec *spec = codec->spec;
|
||||
hda_nid_t nid, mix;
|
||||
int err;
|
||||
char name[32];
|
||||
|
||||
if (!pin)
|
||||
return 0;
|
||||
|
||||
if (pin == 0x17) {
|
||||
/* ALC663 has a mono output pin on 0x17 */
|
||||
nid = alc662_look_for_dac(codec, pin);
|
||||
if (!nid) {
|
||||
char name[32];
|
||||
/* the corresponding DAC is already occupied */
|
||||
if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
|
||||
return 0; /* no way */
|
||||
/* create a switch only */
|
||||
sprintf(name, "%s Playback Switch", pfx);
|
||||
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
|
||||
HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
|
||||
return err;
|
||||
}
|
||||
|
||||
if (alc880_is_fixed_pin(pin)) {
|
||||
nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
|
||||
/* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
|
||||
/* specify the DAC as the extra output */
|
||||
if (!spec->multiout.hp_nid)
|
||||
spec->multiout.hp_nid = nid;
|
||||
else
|
||||
spec->multiout.extra_out_nid[0] = nid;
|
||||
/* control HP volume/switch on the output mixer amp */
|
||||
nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
|
||||
sprintf(name, "%s Playback Volume", pfx);
|
||||
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
|
||||
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
|
||||
if (err < 0)
|
||||
return err;
|
||||
sprintf(name, "%s Playback Switch", pfx);
|
||||
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
|
||||
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
|
||||
if (err < 0)
|
||||
return err;
|
||||
} else if (alc880_is_multi_pin(pin)) {
|
||||
/* set manual connection */
|
||||
/* we have only a switch on HP-out PIN */
|
||||
sprintf(name, "%s Playback Switch", pfx);
|
||||
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
|
||||
return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
|
||||
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
|
||||
}
|
||||
|
||||
mix = alc662_dac_to_mix(codec, pin, nid);
|
||||
if (!mix)
|
||||
return 0;
|
||||
err = alc662_add_vol_ctl(spec, pfx, nid, 3);
|
||||
if (err < 0)
|
||||
return err;
|
||||
}
|
||||
return 0;
|
||||
err = alc662_add_sw_ctl(spec, pfx, mix, 3);
|
||||
if (err < 0)
|
||||
return err;
|
||||
return nid;
|
||||
}
|
||||
|
||||
/* create playback/capture controls for input pins */
|
||||
|
@ -17273,30 +17333,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
|
|||
|
||||
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
|
||||
hda_nid_t nid, int pin_type,
|
||||
int dac_idx)
|
||||
hda_nid_t dac)
|
||||
{
|
||||
int i, num;
|
||||
hda_nid_t srcs[4];
|
||||
|
||||
alc_set_pin_output(codec, nid, pin_type);
|
||||
/* need the manual connection? */
|
||||
if (alc880_is_multi_pin(nid)) {
|
||||
struct alc_spec *spec = codec->spec;
|
||||
int idx = alc880_multi_pin_idx(nid);
|
||||
snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
|
||||
AC_VERB_SET_CONNECT_SEL,
|
||||
alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
|
||||
num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
|
||||
if (num <= 1)
|
||||
return;
|
||||
for (i = 0; i < num; i++) {
|
||||
if (alc662_mix_to_dac(srcs[i]) != dac)
|
||||
continue;
|
||||
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
|
||||
return;
|
||||
}
|
||||
}
|
||||
|
||||
static void alc662_auto_init_multi_out(struct hda_codec *codec)
|
||||
{
|
||||
struct alc_spec *spec = codec->spec;
|
||||
int pin_type = get_pin_type(spec->autocfg.line_out_type);
|
||||
int i;
|
||||
|
||||
for (i = 0; i <= HDA_SIDE; i++) {
|
||||
hda_nid_t nid = spec->autocfg.line_out_pins[i];
|
||||
int pin_type = get_pin_type(spec->autocfg.line_out_type);
|
||||
if (nid)
|
||||
alc662_auto_set_output_and_unmute(codec, nid, pin_type,
|
||||
i);
|
||||
spec->multiout.dac_nids[i]);
|
||||
}
|
||||
}
|
||||
|
||||
|
@ -17306,12 +17371,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
|
|||
hda_nid_t pin;
|
||||
|
||||
pin = spec->autocfg.hp_pins[0];
|
||||
if (pin) /* connect to front */
|
||||
/* use dac 0 */
|
||||
alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
|
||||
if (pin)
|
||||
alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
|
||||
spec->multiout.hp_nid);
|
||||
pin = spec->autocfg.speaker_pins[0];
|
||||
if (pin)
|
||||
alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
|
||||
alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
|
||||
spec->multiout.extra_out_nid[0]);
|
||||
}
|
||||
|
||||
#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
|
||||
|
@ -17349,21 +17415,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
|
|||
if (!spec->autocfg.line_outs)
|
||||
return 0; /* can't find valid BIOS pin config */
|
||||
|
||||
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
|
||||
err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
|
||||
err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = alc662_auto_create_extra_out(spec,
|
||||
err = alc662_auto_create_extra_out(codec,
|
||||
spec->autocfg.speaker_pins[0],
|
||||
"Speaker");
|
||||
if (err < 0)
|
||||
return err;
|
||||
err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
|
||||
if (err)
|
||||
spec->multiout.extra_out_nid[0] = err;
|
||||
err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
|
||||
"Headphone");
|
||||
if (err < 0)
|
||||
return err;
|
||||
if (err)
|
||||
spec->multiout.hp_nid = err;
|
||||
err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
|
||||
if (err < 0)
|
||||
return err;
|
||||
|
|
|
@ -182,8 +182,8 @@ struct sigmatel_jack {
|
|||
|
||||
struct sigmatel_mic_route {
|
||||
hda_nid_t pin;
|
||||
unsigned char mux_idx;
|
||||
unsigned char dmux_idx;
|
||||
signed char mux_idx;
|
||||
signed char dmux_idx;
|
||||
};
|
||||
|
||||
struct sigmatel_spec {
|
||||
|
@ -3469,18 +3469,26 @@ static int set_mic_route(struct hda_codec *codec,
|
|||
break;
|
||||
if (i <= AUTO_PIN_FRONT_MIC) {
|
||||
/* analog pin */
|
||||
mic->dmux_idx = 0;
|
||||
i = get_connection_index(codec, spec->mux_nids[0], pin);
|
||||
if (i < 0)
|
||||
return -1;
|
||||
mic->mux_idx = i;
|
||||
mic->dmux_idx = -1;
|
||||
if (spec->dmux_nids)
|
||||
mic->dmux_idx = get_connection_index(codec,
|
||||
spec->dmux_nids[0],
|
||||
spec->mux_nids[0]);
|
||||
} else if (spec->dmux_nids) {
|
||||
/* digital pin */
|
||||
mic->mux_idx = 0;
|
||||
i = get_connection_index(codec, spec->dmux_nids[0], pin);
|
||||
if (i < 0)
|
||||
return -1;
|
||||
mic->dmux_idx = i;
|
||||
mic->mux_idx = -1;
|
||||
if (spec->mux_nids)
|
||||
mic->mux_idx = get_connection_index(codec,
|
||||
spec->mux_nids[0],
|
||||
spec->dmux_nids[0]);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
@ -4557,11 +4565,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec)
|
|||
mic = &spec->ext_mic;
|
||||
else
|
||||
mic = &spec->int_mic;
|
||||
if (mic->dmux_idx)
|
||||
if (mic->dmux_idx >= 0)
|
||||
snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
|
||||
AC_VERB_SET_CONNECT_SEL,
|
||||
mic->dmux_idx);
|
||||
else
|
||||
if (mic->mux_idx >= 0)
|
||||
snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
|
||||
AC_VERB_SET_CONNECT_SEL,
|
||||
mic->mux_idx);
|
||||
|
|
|
@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
|
|||
.name = "Sony S1XP",
|
||||
.type = AC97_TUNE_INV_EAPD
|
||||
},
|
||||
{
|
||||
.subvendor = 0x104d,
|
||||
.subdevice = 0x81c0,
|
||||
.name = "Sony VAIO VGN-T350P", /*AD1981B*/
|
||||
.type = AC97_TUNE_INV_EAPD
|
||||
},
|
||||
{
|
||||
.subvendor = 0x104d,
|
||||
.subdevice = 0x81c5,
|
||||
.name = "Sony VAIO VGN-B1VP", /*AD1981B*/
|
||||
.type = AC97_TUNE_INV_EAPD
|
||||
},
|
||||
{
|
||||
.subvendor = 0x1043,
|
||||
.subdevice = 0x80f3,
|
||||
|
|
|
@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
|
|||
strlcpy(info.type, "keywest", I2C_NAME_SIZE);
|
||||
info.addr = keywest_ctx->addr;
|
||||
keywest_ctx->client = i2c_new_device(adapter, &info);
|
||||
if (!keywest_ctx->client)
|
||||
return -ENODEV;
|
||||
/*
|
||||
* We know the driver is already loaded, so the device should be
|
||||
* already bound. If not it means binding failed, and then there
|
||||
* is no point in keeping the device instantiated.
|
||||
*/
|
||||
if (!keywest_ctx->client->driver) {
|
||||
i2c_unregister_device(keywest_ctx->client);
|
||||
keywest_ctx->client = NULL;
|
||||
return -ENODEV;
|
||||
}
|
||||
|
||||
/*
|
||||
* Let i2c-core delete that device on driver removal.
|
||||
|
@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = {
|
|||
{ }
|
||||
};
|
||||
|
||||
struct i2c_driver keywest_driver = {
|
||||
static struct i2c_driver keywest_driver = {
|
||||
.driver = {
|
||||
.name = "PMac Keywest Audio",
|
||||
},
|
||||
|
|
|
@ -7,15 +7,6 @@ config SND_BF5XX_I2S
|
|||
mode (supports single stereo In/Out).
|
||||
You will also need to select the audio interfaces to support below.
|
||||
|
||||
config SND_BF5XX_TDM
|
||||
tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
|
||||
depends on (BLACKFIN && SND_SOC)
|
||||
help
|
||||
Say Y or M if you want to add support for codecs attached to
|
||||
the Blackfin SPORT (synchronous serial ports) interface in TDM
|
||||
mode.
|
||||
You will also need to select the audio interfaces to support below.
|
||||
|
||||
config SND_BF5XX_SOC_SSM2602
|
||||
tristate "SoC SSM2602 Audio support for BF52x ezkit"
|
||||
depends on SND_BF5XX_I2S
|
||||
|
@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
|
|||
Enter the GPIO used to control AD73311's SE pin. Acceptable
|
||||
values are 0 to 7
|
||||
|
||||
config SND_BF5XX_TDM
|
||||
tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
|
||||
depends on (BLACKFIN && SND_SOC)
|
||||
help
|
||||
Say Y or M if you want to add support for codecs attached to
|
||||
the Blackfin SPORT (synchronous serial ports) interface in TDM
|
||||
mode.
|
||||
You will also need to select the audio interfaces to support below.
|
||||
|
||||
config SND_BF5XX_SOC_AD1836
|
||||
tristate "SoC AD1836 Audio support for BF5xx"
|
||||
depends on SND_BF5XX_TDM
|
||||
select SND_BF5XX_SOC_TDM
|
||||
select SND_SOC_AD1836
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
|
||||
|
||||
config SND_BF5XX_SOC_AD1938
|
||||
tristate "SoC AD1938 Audio support for Blackfin"
|
||||
depends on SND_BF5XX_TDM
|
||||
select SND_BF5XX_SOC_TDM
|
||||
select SND_SOC_AD1938
|
||||
help
|
||||
Say Y if you want to add support for AD1938 codec on Blackfin.
|
||||
|
||||
config SND_BF5XX_AC97
|
||||
tristate "SoC AC97 Audio for the ADI BF5xx chip"
|
||||
depends on BLACKFIN
|
||||
|
@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
|
|||
Say y if you want AC97 driver to support up to 5.1 channel audio.
|
||||
this mode will consume much more memory for DMA.
|
||||
|
||||
config SND_BF5XX_HAVE_COLD_RESET
|
||||
bool "BOARD has COLD Reset GPIO"
|
||||
depends on SND_BF5XX_AC97
|
||||
default y if BFIN548_EZKIT
|
||||
default n if !BFIN548_EZKIT
|
||||
|
||||
config SND_BF5XX_RESET_GPIO_NUM
|
||||
int "Set a GPIO for cold reset"
|
||||
depends on SND_BF5XX_HAVE_COLD_RESET
|
||||
range 0 159
|
||||
default 19 if BFIN548_EZKIT
|
||||
default 5 if BFIN537_STAMP
|
||||
default 0
|
||||
help
|
||||
Set the correct GPIO for RESET the sound chip.
|
||||
|
||||
config SND_BF5XX_SOC_AD1980
|
||||
tristate "SoC AD1980/1 Audio support for BF5xx"
|
||||
depends on SND_BF5XX_AC97
|
||||
select SND_BF5XX_SOC_AC97
|
||||
select SND_SOC_AD1980
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
|
||||
|
||||
config SND_BF5XX_SOC_SPORT
|
||||
tristate
|
||||
|
||||
|
@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
|
|||
select SND_SOC_AC97_BUS
|
||||
select SND_BF5XX_SOC_SPORT
|
||||
|
||||
config SND_BF5XX_SOC_AD1836
|
||||
tristate "SoC AD1836 Audio support for BF5xx"
|
||||
depends on SND_BF5XX_TDM
|
||||
select SND_BF5XX_SOC_TDM
|
||||
select SND_SOC_AD1836
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
|
||||
|
||||
config SND_BF5XX_SOC_AD1980
|
||||
tristate "SoC AD1980/1 Audio support for BF5xx"
|
||||
depends on SND_BF5XX_AC97
|
||||
select SND_BF5XX_SOC_AC97
|
||||
select SND_SOC_AD1980
|
||||
help
|
||||
Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
|
||||
|
||||
config SND_BF5XX_SOC_AD1938
|
||||
tristate "SoC AD1938 Audio support for Blackfin"
|
||||
depends on SND_BF5XX_TDM
|
||||
select SND_BF5XX_SOC_TDM
|
||||
select SND_SOC_AD1938
|
||||
help
|
||||
Say Y if you want to add support for AD1938 codec on Blackfin.
|
||||
|
||||
config SND_BF5XX_SPORT_NUM
|
||||
int "Set a SPORT for Sound chip"
|
||||
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
|
||||
|
@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
|
|||
default 0
|
||||
help
|
||||
Set the correct SPORT for sound chip.
|
||||
|
||||
config SND_BF5XX_HAVE_COLD_RESET
|
||||
bool "BOARD has COLD Reset GPIO"
|
||||
depends on SND_BF5XX_AC97
|
||||
default y if BFIN548_EZKIT
|
||||
default n if !BFIN548_EZKIT
|
||||
|
||||
config SND_BF5XX_RESET_GPIO_NUM
|
||||
int "Set a GPIO for cold reset"
|
||||
depends on SND_BF5XX_HAVE_COLD_RESET
|
||||
range 0 159
|
||||
default 19 if BFIN548_EZKIT
|
||||
default 5 if BFIN537_STAMP
|
||||
default 0
|
||||
help
|
||||
Set the correct GPIO for RESET the sound chip.
|
||||
|
|
|
@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
|
|||
* 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.
|
||||
* codecs and EMAC using Port G concurrently.
|
||||
*/
|
||||
#ifdef CONFIG_BF527_SPORT0_PORTF
|
||||
#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
|
||||
#else
|
||||
#ifdef CONFIG_BF527_SPORT0_PORTG
|
||||
#define LOCAL_SPORT0_TFS (0)
|
||||
#else
|
||||
#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
|
||||
#endif
|
||||
|
||||
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
|
||||
|
|
|
@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
|
|||
* 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.
|
||||
* codecs and EMAC using Port G concurrently.
|
||||
*/
|
||||
#ifdef CONFIG_BF527_SPORT0_PORTF
|
||||
#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
|
||||
#else
|
||||
#ifdef CONFIG_BF527_SPORT0_PORTG
|
||||
#define LOCAL_SPORT0_TFS (0)
|
||||
#else
|
||||
#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
|
||||
#endif
|
||||
|
||||
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
|
||||
|
|
|
@ -97,22 +97,19 @@ enum {
|
|||
DAVINCI_MCBSP_WORD_32,
|
||||
};
|
||||
|
||||
static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
|
||||
.name = "I2S PCM Stereo out",
|
||||
};
|
||||
|
||||
static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
|
||||
.name = "I2S PCM Stereo in",
|
||||
};
|
||||
|
||||
struct davinci_mcbsp_dev {
|
||||
/*
|
||||
* dma_params must be first because rtd->dai->cpu_dai->private_data
|
||||
* is cast to a pointer of an array of struct davinci_pcm_dma_params in
|
||||
* davinci_pcm_open.
|
||||
*/
|
||||
struct davinci_pcm_dma_params dma_params[2];
|
||||
void __iomem *base;
|
||||
#define MOD_DSP_A 0
|
||||
#define MOD_DSP_B 1
|
||||
int mode;
|
||||
u32 pcr;
|
||||
struct clk *clk;
|
||||
struct davinci_pcm_dma_params *dma_params[2];
|
||||
};
|
||||
|
||||
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
|
||||
|
@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
|
|||
toggle_clock(dev, playback);
|
||||
}
|
||||
|
||||
static int davinci_i2s_startup(struct snd_pcm_substream *substream,
|
||||
struct snd_soc_dai *cpu_dai)
|
||||
{
|
||||
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
|
||||
cpu_dai->dma_data = dev->dma_params[substream->stream];
|
||||
return 0;
|
||||
}
|
||||
|
||||
#define DEFAULT_BITPERSAMPLE 16
|
||||
|
||||
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
|
||||
|
@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
|
|||
struct snd_pcm_hw_params *params,
|
||||
struct snd_soc_dai *dai)
|
||||
{
|
||||
struct davinci_pcm_dma_params *dma_params = dai->dma_data;
|
||||
struct davinci_mcbsp_dev *dev = dai->private_data;
|
||||
struct davinci_pcm_dma_params *dma_params =
|
||||
&dev->dma_params[substream->stream];
|
||||
struct snd_interval *i = NULL;
|
||||
int mcbsp_word_length;
|
||||
unsigned int rcr, xcr, srgr;
|
||||
|
@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
|
|||
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
|
||||
|
||||
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
|
||||
.startup = davinci_i2s_startup,
|
||||
.shutdown = davinci_i2s_shutdown,
|
||||
.prepare = davinci_i2s_prepare,
|
||||
.trigger = davinci_i2s_trigger,
|
||||
|
@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
|
|||
|
||||
dev->base = (void __iomem *)IO_ADDRESS(mem->start);
|
||||
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
|
||||
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
|
||||
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
|
||||
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
|
||||
|
||||
/* first TX, then RX */
|
||||
|
@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
|
|||
ret = -ENXIO;
|
||||
goto err_free_mem;
|
||||
}
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
|
||||
|
||||
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
|
||||
if (!res) {
|
||||
|
@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
|
|||
ret = -ENXIO;
|
||||
goto err_free_mem;
|
||||
}
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
|
||||
|
||||
davinci_i2s_dai.private_data = dev;
|
||||
ret = snd_soc_register_dai(&davinci_i2s_dai);
|
||||
|
|
|
@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
|
|||
printk(KERN_ERR "GBLCTL write error\n");
|
||||
}
|
||||
|
||||
static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
|
||||
struct snd_soc_dai *cpu_dai)
|
||||
{
|
||||
struct davinci_audio_dev *dev = cpu_dai->private_data;
|
||||
cpu_dai->dma_data = dev->dma_params[substream->stream];
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void mcasp_start_rx(struct davinci_audio_dev *dev)
|
||||
{
|
||||
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
|
||||
|
@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
|
|||
|
||||
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
|
||||
{
|
||||
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
|
||||
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
||||
if (dev->txnumevt) /* enable FIFO */
|
||||
mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
|
||||
FIFO_ENABLE);
|
||||
mcasp_start_tx(dev);
|
||||
else
|
||||
} else {
|
||||
if (dev->rxnumevt) /* enable FIFO */
|
||||
mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
|
||||
FIFO_ENABLE);
|
||||
mcasp_start_rx(dev);
|
||||
|
||||
/* enable FIFO */
|
||||
if (dev->txnumevt)
|
||||
mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
|
||||
|
||||
if (dev->rxnumevt)
|
||||
mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
|
||||
}
|
||||
}
|
||||
|
||||
static void mcasp_stop_rx(struct davinci_audio_dev *dev)
|
||||
|
@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
|
|||
|
||||
static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
|
||||
{
|
||||
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
|
||||
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
||||
if (dev->txnumevt) /* disable FIFO */
|
||||
mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
|
||||
FIFO_ENABLE);
|
||||
mcasp_stop_tx(dev);
|
||||
else
|
||||
} else {
|
||||
if (dev->rxnumevt) /* disable FIFO */
|
||||
mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
|
||||
FIFO_ENABLE);
|
||||
mcasp_stop_rx(dev);
|
||||
|
||||
/* disable FIFO */
|
||||
if (dev->txnumevt)
|
||||
mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
|
||||
|
||||
if (dev->rxnumevt)
|
||||
mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
|
||||
}
|
||||
}
|
||||
|
||||
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
|
||||
|
@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
|
|||
{
|
||||
struct davinci_audio_dev *dev = cpu_dai->private_data;
|
||||
struct davinci_pcm_dma_params *dma_params =
|
||||
dev->dma_params[substream->stream];
|
||||
&dev->dma_params[substream->stream];
|
||||
int word_length;
|
||||
u8 numevt;
|
||||
|
||||
|
@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
|
|||
}
|
||||
|
||||
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
|
||||
.startup = davinci_mcasp_startup,
|
||||
.trigger = davinci_mcasp_trigger,
|
||||
.hw_params = davinci_mcasp_hw_params,
|
||||
.set_fmt = davinci_mcasp_set_dai_fmt,
|
||||
|
@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
|
|||
struct resource *mem, *ioarea, *res;
|
||||
struct snd_platform_data *pdata;
|
||||
struct davinci_audio_dev *dev;
|
||||
int count = 0;
|
||||
int ret = 0;
|
||||
|
||||
dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
|
||||
if (!dev)
|
||||
return -ENOMEM;
|
||||
|
||||
dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
|
||||
GFP_KERNEL);
|
||||
if (!dma_data) {
|
||||
ret = -ENOMEM;
|
||||
goto err_release_dev;
|
||||
}
|
||||
|
||||
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
|
||||
if (!mem) {
|
||||
dev_err(&pdev->dev, "no mem resource?\n");
|
||||
|
@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
|
|||
dev->txnumevt = pdata->txnumevt;
|
||||
dev->rxnumevt = pdata->rxnumevt;
|
||||
|
||||
dma_data[count].name = "I2S PCM Stereo out";
|
||||
dma_data[count].eventq_no = pdata->eventq_no;
|
||||
dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
|
||||
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
|
||||
dma_data->eventq_no = pdata->eventq_no;
|
||||
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
|
||||
io_v2p(dev->base));
|
||||
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
|
||||
|
||||
/* first TX, then RX */
|
||||
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
|
||||
|
@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
|
|||
goto err_release_region;
|
||||
}
|
||||
|
||||
dma_data[count].channel = res->start;
|
||||
count++;
|
||||
dma_data[count].name = "I2S PCM Stereo in";
|
||||
dma_data[count].eventq_no = pdata->eventq_no;
|
||||
dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
|
||||
dma_data->channel = res->start;
|
||||
|
||||
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
|
||||
dma_data->eventq_no = pdata->eventq_no;
|
||||
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
|
||||
io_v2p(dev->base));
|
||||
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
|
||||
|
||||
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
|
||||
if (!res) {
|
||||
|
@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
|
|||
goto err_release_region;
|
||||
}
|
||||
|
||||
dma_data[count].channel = res->start;
|
||||
dma_data->channel = res->start;
|
||||
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
|
||||
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
|
||||
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
|
||||
|
@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
|
|||
err_release_region:
|
||||
release_mem_region(mem->start, (mem->end - mem->start) + 1);
|
||||
err_release_data:
|
||||
kfree(dma_data);
|
||||
err_release_dev:
|
||||
kfree(dev);
|
||||
|
||||
return ret;
|
||||
|
@ -946,7 +925,6 @@ err_release_dev:
|
|||
static int davinci_mcasp_remove(struct platform_device *pdev)
|
||||
{
|
||||
struct snd_platform_data *pdata = pdev->dev.platform_data;
|
||||
struct davinci_pcm_dma_params *dma_data;
|
||||
struct davinci_audio_dev *dev;
|
||||
struct resource *mem;
|
||||
|
||||
|
@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
|
|||
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
|
||||
release_mem_region(mem->start, (mem->end - mem->start) + 1);
|
||||
|
||||
dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
|
||||
kfree(dma_data);
|
||||
kfree(dev);
|
||||
|
||||
return 0;
|
||||
|
|
|
@ -39,10 +39,15 @@ enum {
|
|||
};
|
||||
|
||||
struct davinci_audio_dev {
|
||||
/*
|
||||
* dma_params must be first because rtd->dai->cpu_dai->private_data
|
||||
* is cast to a pointer of an array of struct davinci_pcm_dma_params in
|
||||
* davinci_pcm_open.
|
||||
*/
|
||||
struct davinci_pcm_dma_params dma_params[2];
|
||||
void __iomem *base;
|
||||
int sample_rate;
|
||||
struct clk *clk;
|
||||
struct davinci_pcm_dma_params *dma_params[2];
|
||||
unsigned int codec_fmt;
|
||||
|
||||
/* McASP specific data */
|
||||
|
|
|
@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
|
|||
static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
|
||||
{
|
||||
struct davinci_runtime_data *prtd = substream->runtime->private_data;
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
|
||||
struct edmacc_param p_ram;
|
||||
int ret;
|
||||
|
||||
if (!dma_data)
|
||||
return -ENODEV;
|
||||
|
||||
prtd->params = dma_data;
|
||||
|
||||
/* Request master DMA channel */
|
||||
ret = edma_alloc_channel(prtd->params->channel,
|
||||
davinci_pcm_dma_irq, substream,
|
||||
|
@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
|
|||
struct snd_pcm_runtime *runtime = substream->runtime;
|
||||
struct davinci_runtime_data *prtd;
|
||||
int ret = 0;
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
|
||||
struct davinci_pcm_dma_params *params = &pa[substream->stream];
|
||||
if (!params)
|
||||
return -ENODEV;
|
||||
|
||||
snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
|
||||
/* ensure that buffer size is a multiple of period size */
|
||||
|
@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
|
|||
return -ENOMEM;
|
||||
|
||||
spin_lock_init(&prtd->lock);
|
||||
prtd->params = params;
|
||||
|
||||
runtime->private_data = prtd;
|
||||
|
||||
|
|
|
@ -17,7 +17,6 @@
|
|||
|
||||
|
||||
struct davinci_pcm_dma_params {
|
||||
char *name; /* stream identifier */
|
||||
int channel; /* sync dma channel ID */
|
||||
unsigned short acnt;
|
||||
dma_addr_t dma_addr; /* device physical address for DMA */
|
||||
|
|
|
@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
|
|||
|
||||
config SND_PXA2XX_SOC_IMOTE2
|
||||
tristate "SoC Audio support for IMote 2"
|
||||
depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
|
||||
depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
|
||||
select SND_PXA2XX_SOC_I2S
|
||||
select SND_SOC_WM8940
|
||||
help
|
||||
|
|
|
@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = {
|
|||
* build a feature control
|
||||
*/
|
||||
|
||||
static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
|
||||
{
|
||||
return strlcat(kctl->id.name, str, sizeof(kctl->id.name));
|
||||
}
|
||||
|
||||
static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
|
||||
unsigned int ctl_mask, int control,
|
||||
struct usb_audio_term *iterm, int unitid)
|
||||
|
@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
|
|||
*/
|
||||
if (! mapped_name && ! (state->oterm.type >> 16)) {
|
||||
if ((state->oterm.type & 0xff00) == 0x0100) {
|
||||
len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
|
||||
len = append_ctl_name(kctl, " Capture");
|
||||
} else {
|
||||
len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
|
||||
len = append_ctl_name(kctl, " Playback");
|
||||
}
|
||||
}
|
||||
strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
|
||||
sizeof(kctl->id.name));
|
||||
append_ctl_name(kctl, control == USB_FEATURE_MUTE ?
|
||||
" Switch" : " Volume");
|
||||
if (control == USB_FEATURE_VOLUME) {
|
||||
kctl->tlv.c = mixer_vol_tlv;
|
||||
kctl->vd[0].access |=
|
||||
|
@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
|
|||
len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
|
||||
if (! len)
|
||||
len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
|
||||
strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
|
||||
append_ctl_name(kctl, " Volume");
|
||||
|
||||
snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
|
||||
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
|
||||
|
@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned
|
|||
if (! len)
|
||||
strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
|
||||
}
|
||||
strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
|
||||
strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
|
||||
append_ctl_name(kctl, " ");
|
||||
append_ctl_name(kctl, valinfo->suffix);
|
||||
|
||||
snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
|
||||
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
|
||||
|
@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi
|
|||
strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
|
||||
|
||||
if ((state->oterm.type & 0xff00) == 0x0100)
|
||||
strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
|
||||
append_ctl_name(kctl, " Capture Source");
|
||||
else
|
||||
strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
|
||||
append_ctl_name(kctl, " Playback Source");
|
||||
}
|
||||
|
||||
snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
|
||||
|
|
Loading…
Reference in New Issue