ALSA - hda: Add support for link audio time reporting
The HDA controller from SKL onwards support additional timestamp reporting of the link time. The link time is read from HW registers and converted to audio values. Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com> Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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@ -27,6 +27,12 @@
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#include <linux/module.h>
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#include <linux/pm_runtime.h>
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#include <linux/slab.h>
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#ifdef CONFIG_X86
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/* for art-tsc conversion */
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#include <asm/tsc.h>
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#endif
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#include <sound/core.h>
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#include <sound/initval.h>
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#include "hda_controller.h"
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@ -337,12 +343,173 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
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azx_get_position(chip, azx_dev));
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}
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/*
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* azx_scale64: Scale base by mult/div while not overflowing sanely
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*
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* Derived from scale64_check_overflow in kernel/time/timekeeping.c
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*
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* The tmestamps for a 48Khz stream can overflow after (2^64/10^9)/48K which
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* is about 384307 ie ~4.5 days.
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*
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* This scales the calculation so that overflow will happen but after 2^64 /
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* 48000 secs, which is pretty large!
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*
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* In caln below:
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* base may overflow, but since there isn’t any additional division
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* performed on base it’s OK
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* rem can’t overflow because both are 32-bit values
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*/
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#ifdef CONFIG_X86
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static u64 azx_scale64(u64 base, u32 num, u32 den)
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{
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u64 rem;
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rem = do_div(base, den);
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base *= num;
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rem *= num;
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do_div(rem, den);
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return base + rem;
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}
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static int azx_get_sync_time(ktime_t *device,
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struct system_counterval_t *system, void *ctx)
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{
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struct snd_pcm_substream *substream = ctx;
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struct azx_dev *azx_dev = get_azx_dev(substream);
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struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
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struct azx *chip = apcm->chip;
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struct snd_pcm_runtime *runtime;
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u64 ll_counter, ll_counter_l, ll_counter_h;
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u64 tsc_counter, tsc_counter_l, tsc_counter_h;
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u32 wallclk_ctr, wallclk_cycles;
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bool direction;
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u32 dma_select;
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u32 timeout = 200;
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u32 retry_count = 0;
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runtime = substream->runtime;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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direction = 1;
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else
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direction = 0;
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/* 0th stream tag is not used, so DMA ch 0 is for 1st stream tag */
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do {
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timeout = 100;
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dma_select = (direction << GTSCC_CDMAS_DMA_DIR_SHIFT) |
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(azx_dev->core.stream_tag - 1);
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snd_hdac_chip_writel(azx_bus(chip), GTSCC, dma_select);
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/* Enable the capture */
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snd_hdac_chip_updatel(azx_bus(chip), GTSCC, 0, GTSCC_TSCCI_MASK);
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while (timeout) {
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if (snd_hdac_chip_readl(azx_bus(chip), GTSCC) &
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GTSCC_TSCCD_MASK)
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break;
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timeout--;
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}
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if (!timeout) {
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dev_err(chip->card->dev, "GTSCC capture Timedout!\n");
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return -EIO;
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}
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/* Read wall clock counter */
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wallclk_ctr = snd_hdac_chip_readl(azx_bus(chip), WALFCC);
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/* Read TSC counter */
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tsc_counter_l = snd_hdac_chip_readl(azx_bus(chip), TSCCL);
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tsc_counter_h = snd_hdac_chip_readl(azx_bus(chip), TSCCU);
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/* Read Link counter */
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ll_counter_l = snd_hdac_chip_readl(azx_bus(chip), LLPCL);
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ll_counter_h = snd_hdac_chip_readl(azx_bus(chip), LLPCU);
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/* Ack: registers read done */
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snd_hdac_chip_writel(azx_bus(chip), GTSCC, GTSCC_TSCCD_SHIFT);
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tsc_counter = (tsc_counter_h << TSCCU_CCU_SHIFT) |
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tsc_counter_l;
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ll_counter = (ll_counter_h << LLPC_CCU_SHIFT) | ll_counter_l;
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wallclk_cycles = wallclk_ctr & WALFCC_CIF_MASK;
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/*
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* An error occurs near frame "rollover". The clocks in
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* frame value indicates whether this error may have
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* occurred. Here we use the value of 10 i.e.,
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* HDA_MAX_CYCLE_OFFSET
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*/
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if (wallclk_cycles < HDA_MAX_CYCLE_VALUE - HDA_MAX_CYCLE_OFFSET
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&& wallclk_cycles > HDA_MAX_CYCLE_OFFSET)
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break;
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/*
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* Sleep before we read again, else we may again get
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* value near to MAX_CYCLE. Try to sleep for different
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* amount of time so we dont hit the same number again
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*/
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udelay(retry_count++);
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} while (retry_count != HDA_MAX_CYCLE_READ_RETRY);
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if (retry_count == HDA_MAX_CYCLE_READ_RETRY) {
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dev_err_ratelimited(chip->card->dev,
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"Error in WALFCC cycle count\n");
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return -EIO;
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}
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*device = ns_to_ktime(azx_scale64(ll_counter,
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NSEC_PER_SEC, runtime->rate));
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*device = ktime_add_ns(*device, (wallclk_cycles * NSEC_PER_SEC) /
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((HDA_MAX_CYCLE_VALUE + 1) * runtime->rate));
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*system = convert_art_to_tsc(tsc_counter);
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return 0;
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}
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#else
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static int azx_get_sync_time(ktime_t *device,
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struct system_counterval_t *system, void *ctx)
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{
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return -ENXIO;
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}
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#endif
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static int azx_get_crosststamp(struct snd_pcm_substream *substream,
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struct system_device_crosststamp *xtstamp)
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{
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return get_device_system_crosststamp(azx_get_sync_time,
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substream, NULL, xtstamp);
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}
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static inline bool is_link_time_supported(struct snd_pcm_runtime *runtime,
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struct snd_pcm_audio_tstamp_config *ts)
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{
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if (runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME)
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if (ts->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)
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return true;
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return false;
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}
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static int azx_get_time_info(struct snd_pcm_substream *substream,
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struct timespec *system_ts, struct timespec *audio_ts,
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struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
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struct snd_pcm_audio_tstamp_report *audio_tstamp_report)
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{
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struct azx_dev *azx_dev = get_azx_dev(substream);
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct system_device_crosststamp xtstamp;
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int ret;
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u64 nsec;
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if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) &&
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@ -361,8 +528,37 @@ static int azx_get_time_info(struct snd_pcm_substream *substream,
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audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */
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audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */
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} else
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} else if (is_link_time_supported(runtime, audio_tstamp_config)) {
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ret = azx_get_crosststamp(substream, &xtstamp);
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if (ret)
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return ret;
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switch (runtime->tstamp_type) {
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case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC:
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return -EINVAL;
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case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW:
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*system_ts = ktime_to_timespec(xtstamp.sys_monoraw);
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break;
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default:
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*system_ts = ktime_to_timespec(xtstamp.sys_realtime);
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break;
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}
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*audio_ts = ktime_to_timespec(xtstamp.device);
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audio_tstamp_report->actual_type =
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED;
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audio_tstamp_report->accuracy_report = 1;
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/* 24 MHz WallClock == 42ns resolution */
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audio_tstamp_report->accuracy = 42;
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} else {
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audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT;
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}
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return 0;
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}
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