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Commit af8f9372 authored by Linus Torvalds's avatar Linus Torvalds
Browse files

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: hda - Set function_id only on FG nodes
  ALSA: MAINTAINERS - Update SOUND
  ALSA: emu10k1 - off by 1 in snd_emu10k1_wait()
  ASoC: OMAP: Fix FS polarity in OSK5912 machine driver
  ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver
  ASoC: Fix include build error in s3c2412-i2s.c
  ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes
  ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate
  ASoC: Fix jive_wm8750.c build problems
  ASoC: pxa-ssp: allow setting of dai format 0
  ALSA: hda - Add upper-limit of mixer amp for AD1884A-laptop model, too
  ALSA: hda - Fix headphone-detection on some machines with STAC/IDT codecs
  ALSA: Intel8x0: Add hp_only quirk for SSID 0x1028016a (Dell Inspiron 8600)
  ALSA: Intel8x0: Remove conflicting quirk for SSID 0x103c0934
  ALSA: hda_intel.c - Consolidate bitfields
parents 091ccb00 d6aa764e

MAINTAINERS

+5 −0
Original line number Diff line number Diff line
@@ -5235,7 +5235,12 @@ M: perex@perex.cz 
P:	Takashi Iwai

M:	tiwai@suse.de

L:	alsa-devel@alsa-project.org (subscribers-only)

W:	http://www.alsa-project.org/

T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git

T:	git git://git.alsa-project.org/alsa-kernel.git

S:	Maintained

F:	Documentation/sound/

F:	include/sound/

F:	sound/



SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)

sound/pci/emu10k1/io.c

+1 −1
Original line number Diff line number Diff line
@@ -490,7 +490,7 @@ void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait) 
			if (newtime != curtime)

				break;

		}

		if (count >= 16384)

		if (count > 16384)

			break;

		curtime = newtime;

	}

sound/pci/hda/hda_codec.c

+5 −3
Original line number Diff line number Diff line
@@ -642,19 +642,21 @@ static int get_codec_name(struct hda_codec *codec) 
 */

static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)

{

	int i, total_nodes;

	int i, total_nodes, function_id;

	hda_nid_t nid;



	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);

	for (i = 0; i < total_nodes; i++, nid++) {

		codec->function_id = snd_hda_param_read(codec, nid,

		function_id = snd_hda_param_read(codec, nid,

						AC_PAR_FUNCTION_TYPE) & 0xff;

		switch (codec->function_id) {

		switch (function_id) {

		case AC_GRP_AUDIO_FUNCTION:

			codec->afg = nid;

			codec->function_id = function_id;

			break;

		case AC_GRP_MODEM_FUNCTION:

			codec->mfg = nid;

			codec->function_id = function_id;

			break;

		default:

			break;

sound/pci/hda/hda_intel.c

+1 −1
Original line number Diff line number Diff line
@@ -312,7 +312,6 @@ struct azx_dev { 
	unsigned int period_bytes; /* size of the period in bytes */

	unsigned int frags;	/* number for period in the play buffer */

	unsigned int fifo_size;	/* FIFO size */

	unsigned int start_flag: 1;	/* stream full start flag */

	unsigned long start_jiffies;	/* start + minimum jiffies */

	unsigned long min_jiffies;	/* minimum jiffies before position is valid */
@@ -333,6 +332,7 @@ struct azx_dev { 
	unsigned int opened :1;

	unsigned int running :1;

	unsigned int irq_pending :1;

	unsigned int start_flag: 1;	/* stream full start flag */

	/*

	 * For VIA:

	 *  A flag to ensure DMA position is 0

sound/pci/hda/patch_analog.c

+8 −0
Original line number Diff line number Diff line
@@ -3977,6 +3977,14 @@ static int patch_ad1884a(struct hda_codec *codec) 
		spec->input_mux = &ad1884a_laptop_capture_source;

		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;

		codec->patch_ops.init = ad1884a_hp_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;

	case AD1884A_MOBILE:

		spec->mixers[0] = ad1884a_mobile_mixers;