Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

1 files changed, 9 insertions, 6 deletions

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2da8ad75fd96..1a033177b83f 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -82,6 +82,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 };
 static int nrpacks = 8;		/* max. number of packets per urb */
 static int device_setup[SNDRV_CARDS]; /* device parameter for this card */
 static bool ignore_ctl_error;
+static bool autoclock = true;
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
@@ -100,6 +101,8 @@ MODULE_PARM_DESC(device_setup, "Specific device setup (if needed).");
 module_param(ignore_ctl_error, bool, 0444);
 MODULE_PARM_DESC(ignore_ctl_error,
 		 "Ignore errors from USB controller for mixer interfaces.");
+module_param(autoclock, bool, 0444);
+MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes).");
 
 /*
  * we keep the snd_usb_audio_t instances by ourselves for merging
@@ -354,6 +357,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 	chip->card = card;
 	chip->setup = device_setup[idx];
 	chip->nrpacks = nrpacks;
+	chip->autoclock = autoclock;
 	chip->probing = 1;
 
 	chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
@@ -627,7 +631,9 @@ int snd_usb_autoresume(struct snd_usb_audio *chip)
 	int err = -ENODEV;
 
 	down_read(&chip->shutdown_rwsem);
-	if (!chip->shutdown && !chip->probing)
+	if (chip->probing)
+		err = 0;
+	else if (!chip->shutdown)
 		err = usb_autopm_get_interface(chip->pm_intf);
 	up_read(&chip->shutdown_rwsem);
 
@@ -645,7 +651,6 @@ void snd_usb_autosuspend(struct snd_usb_audio *chip)
 static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 {
 	struct snd_usb_audio *chip = usb_get_intfdata(intf);
-	struct list_head *p;
 	struct snd_usb_stream *as;
 	struct usb_mixer_interface *mixer;
 
@@ -655,8 +660,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 	if (!PMSG_IS_AUTO(message)) {
 		snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
 		if (!chip->num_suspended_intf++) {
-			list_for_each(p, &chip->pcm_list) {
-				as = list_entry(p, struct snd_usb_stream, list);
+			list_for_each_entry(as, &chip->pcm_list, list) {
 				snd_pcm_suspend_all(as->pcm);
 				as->substream[0].need_setup_ep =
 					as->substream[1].need_setup_ep = true;
@@ -716,8 +720,7 @@ static struct usb_device_id usb_audio_ids [] = {
       .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL },
     { }						/* Terminating entry */
 };
-
-MODULE_DEVICE_TABLE (usb, usb_audio_ids);
+MODULE_DEVICE_TABLE(usb, usb_audio_ids);
 
 /*
  * entry point for linux usb interface