Merge tag 'sound-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It's been relatively calm in this cycle from the feature POV, but
  there were lots of cleanup works in the wide-range of code for
  converting with the auto-cleanup macros like guard().

  The mostly user-visible changes are the support of a couple of new
  compress-offload API extensions, and the support of new ASoC codec /
  platform drivers as well as USB-audio quirks.

  Here we go with some highlights:

  Core:
   - Compress-offload API extension for 64bit timestamp support
   - Compress-offload API extension for OPUS codec support
   - Workaround for PCM locking issue with PREEMPT_RT and softirq
   - KCSAN warning fix for ALSA sequencer core

  ASoC:
   - Continued cleanup works for ASoC core APIs
   - Lots of cleanups and conversions of DT bindings
   - Substantial maintainance work on the Intel AVS drivers
   - Support for Qualcomm Glymur and PM4125, Realtek RT1321, Shanghai
     FourSemi FS2104/5S, Texas Instruments PCM1754 and TAS2783A
   - Remove support for TI WL1273 for old Nokia systems

  USB-audio:
   - Support for Tascam US-144mkII, Presonus S1824c support
   - More flexible quirk option handling
   - Fix for USB MIDI timer bug triggered by fuzzer

  Others:
   - A large series of cleanups with guard() & co macros over (non-ASoC)
     sound drivers (PCI, ISA, HD-audio, USB-audio, drivers, etc)
   - TAS5825 HD-audio side-codec support"

* tag 'sound-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (454 commits)
  ALSA: usb-audio: don't hardcode gain for output channel of Presonus Studio
  ALSA: usb-audio: add the initial mix for Presonus Studio 1824c
  ALSA: doc: improved docs about quirk_flags in snd-usb-audio
  ALSA: usb-audio: make param quirk_flags change-able in runtime
  ALSA: usb-audio: improve module param quirk_flags
  ALSA: usb-audio: add two-way convert between name and bit for QUIRK_FLAG_*
  ALSA: usb-audio: fix race condition to UAF in snd_usbmidi_free
  ALSA: usb-audio: add mono main switch to Presonus S1824c
  ALSA: compress: document 'chan_map' member in snd_dec_opus
  ASoC: cs35l56: Add support for CS35L56 B2 silicon
  ASoC: cs35l56: Set fw_regs table after getting REVID
  ALSA: hda/realtek: Add quirk for HP Spectre 14t-ea100
  ASoc: tas2783A: Fix an error code in probe()
  ASoC: tlv320aic3x: Fix class-D initialization for tlv320aic3007
  ASoC: qcom: sc8280xp: use sa8775p/ subdir for QCS9100 / QCS9075
  ASoC: stm32: sai: manage context in set_sysclk callback
  ASoC: renesas: msiof: ignore 1st FSERR
  ASoC: renesas: msiof: Add note for The possibility of R/L opposite Capture
  ASoC: renesas: msiof: setup both (Playback/Capture) in the same time
  ASoC: renesas: msiof: tidyup DMAC stop timing
  ...

1 files changed, 5 insertions, 4 deletions

diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c
index 8bb27f86eb65..2646c4632ca1 100644
--- a/sound/soc/intel/atom/sst/sst_drv_interface.c
+++ b/sound/soc/intel/atom/sst/sst_drv_interface.c
@@ -326,7 +326,7 @@ static int sst_cdev_stream_partial_drain(struct device *dev,
 }
 
 static int sst_cdev_tstamp(struct device *dev, unsigned int str_id,
-		struct snd_compr_tstamp *tstamp)
+			   struct snd_compr_tstamp64 *tstamp)
 {
 	struct snd_sst_tstamp fw_tstamp = {0,};
 	struct stream_info *stream;
@@ -349,10 +349,11 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id,
 			(u64)stream->num_ch * SST_GET_BYTES_PER_SAMPLE(24));
 	tstamp->sampling_rate = fw_tstamp.sampling_frequency;
 
-	dev_dbg(dev, "PCM  = %u\n", tstamp->pcm_io_frames);
-	dev_dbg(dev, "Ptr Query on strid = %d  copied_total %d, decodec %d\n",
+	dev_dbg(dev, "PCM  = %llu\n", tstamp->pcm_io_frames);
+	dev_dbg(dev,
+		"Ptr Query on strid = %d  copied_total %llu, decodec %llu\n",
 		str_id, tstamp->copied_total, tstamp->pcm_frames);
-	dev_dbg(dev, "rendered %d\n", tstamp->pcm_io_frames);
+	dev_dbg(dev, "rendered %llu\n", tstamp->pcm_io_frames);
 
 	return 0;
 }