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authorLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
commit4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch)
treecafffb586c60dddfb04b8619fa1ae0e859600de7 /sound/pci/hda/patch_via.c
parentf7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff)
parent60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff)
downloadlinux-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.xz
Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "It was a busy development cycle at this time, as you can see a wide range of changes in diffstat. There are no big changes but many refactoring and improvements. Here we go some highlights: ALSA core: - Procfs codes were cleaned up to use seq_file - Procfs can be opt out via Kconfig (only for EXPERT) - Two types of jack API were unified finally; now both kctl and input jack devs are handled via a single function call. HD-audio: - Continued code restructuring for the future ASoC driver; now HDA controller driver is split to a core helper module. - Preliminary codes for Skylake audio support in HDA core. - Proper i915 gfx power well management for SKL & co - Enabled runtime PM as default for Intel HDMI/DP codecs - Newer Tegra chip supports - More quirks for Dell headsets, Alienware (with CA0132), etc. - A couple of DRM ELD helper API functions ASoC: - Support for loading ASoC topology maps from firmware, intended to be used to allow self-describing DSP firmware images to be built which can map controls added by the DSP to userspace without the kernel needing to know about individual DSP firmwares - Lots of refactoring to avoid direct access to snd_soc_codec where it's not needed supporting future refactoring - Big refactoring, cleanup and enhancement for the Wolfson ADSP driver - Cleanup series for TI TAS2552 and R-CAR drivers - Fixes and improvements on RT56xx codecs - Support for TI TAS571x power amplifiers - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs - Support for x86 systems with RT5650 and Qualcomm Storm - Support for Mediatek AFE (Audio Front End) unit - Other various small fixes to ASoC codec drivers Firewire: - Enhanced to allow non-blocking streams to use timestamp synchronization - Improve support for DM1500 and BeBoBv3 Misc: - Cleanup of old pci API functions over all PCI sound drivers - Fix long-standing regression of the old powermac i2c setup" * tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits) ALSA: pcm: Fix pcm_class sysfs output ALSA: hda-beep: Update authors dead email address ASoC: wm_adsp: Move DSP Rate controls into the codec ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case ALSA: hda: provide default bus io ops extended hdac ALSA: hda: add hda link cleanup routine ALSA: hda: add hdac_ext stream creation and cleanup routines ASoC: rsrc-card: remove unused ret ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core ASoC: mediatek: Add machine driver for rt5650 rt5676 codec ASoC: mediatek: Add machine driver for MAX98090 codec ASoC: mediatek: Add AFE platform driver ASoC: rsnd: remove io from rsnd_mod ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working() ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr() ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA ...
Diffstat (limited to 'sound/pci/hda/patch_via.c')
-rw-r--r--sound/pci/hda/patch_via.c21
1 files changed, 6 insertions, 15 deletions
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index bab6c04932aa..0521be8d46a8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -117,6 +117,8 @@ static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream,
int action);
+static const struct hda_codec_ops via_patch_ops; /* defined below */
+
static struct via_spec *via_new_spec(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -137,6 +139,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
codec->power_save_node = 1;
spec->gen.power_down_unused = 1;
+ codec->patch_ops = via_patch_ops;
return spec;
}
@@ -481,7 +484,6 @@ static const struct hda_codec_ops via_patch_ops = {
.init = via_init,
.free = via_free,
.unsol_event = snd_hda_jack_unsol_event,
- .stream_pm = snd_hda_gen_stream_pm,
#ifdef CONFIG_PM
.suspend = via_suspend,
.resume = via_resume,
@@ -661,6 +663,9 @@ static int patch_vt1708(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ /* override some patch_ops */
+ codec->patch_ops.build_controls = vt1708_build_controls;
+ codec->patch_ops.build_pcms = vt1708_build_pcms;
spec->gen.mixer_nid = 0x17;
/* set jackpoll_interval while parsing the codec */
@@ -689,10 +694,6 @@ static int patch_vt1708(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs;
- codec->patch_ops = via_patch_ops;
- codec->patch_ops.build_controls = vt1708_build_controls;
- codec->patch_ops.build_pcms = vt1708_build_pcms;
-
/* clear jackpoll_interval again; it's set dynamically */
codec->jackpoll_interval = 0;
@@ -717,8 +718,6 @@ static int patch_vt1709(struct hda_codec *codec)
return err;
}
- codec->patch_ops = via_patch_ops;
-
return 0;
}
@@ -745,7 +744,6 @@ static int patch_vt1708B(struct hda_codec *codec)
return err;
}
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -810,7 +808,6 @@ static int patch_vt1708S(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -852,7 +849,6 @@ static int patch_vt1702(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -925,7 +921,6 @@ static int patch_vt1718S(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -1025,7 +1020,6 @@ static int patch_vt1716S(struct hda_codec *codec)
spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer;
spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -1133,7 +1127,6 @@ static int patch_vt2002P(struct hda_codec *codec)
else
spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -1172,7 +1165,6 @@ static int patch_vt1812(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}
@@ -1210,7 +1202,6 @@ static int patch_vt3476(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs;
- codec->patch_ops = via_patch_ops;
return 0;
}