[alsa-devel] Applied "ALSA - hda: Add support for link audio time reporting" to the asoc tree

Mark Brown broonie at kernel.org
Tue Aug 9 14:44:09 CEST 2016


The patch

   ALSA - hda: Add support for link audio time reporting

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From bfcba288b97f10c22fb84f0898ebfb6b468b80ea Mon Sep 17 00:00:00 2001
From: Guneshwor Singh <guneshwor.o.singh at intel.com>
Date: Thu, 4 Aug 2016 15:46:04 +0530
Subject: [PATCH] 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 at intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah at intel.com>
Signed-off-by: Takashi Iwai <tiwai at suse.de>
---
 sound/pci/hda/hda_controller.c | 198 ++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 197 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 1567fe209e01..2ad3b447483f 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -27,6 +27,12 @@
 #include <linux/module.h>
 #include <linux/pm_runtime.h>
 #include <linux/slab.h>
+
+#ifdef CONFIG_X86
+/* for art-tsc conversion */
+#include <asm/tsc.h>
+#endif
+
 #include <sound/core.h>
 #include <sound/initval.h>
 #include "hda_controller.h"
@@ -337,12 +343,173 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
 			       azx_get_position(chip, azx_dev));
 }
 
+/*
+ * azx_scale64: Scale base by mult/div while not overflowing sanely
+ *
+ * Derived from scale64_check_overflow in kernel/time/timekeeping.c
+ *
+ * The tmestamps for a 48Khz stream can overflow after (2^64/10^9)/48K which
+ * is about 384307 ie ~4.5 days.
+ *
+ * This scales the calculation so that overflow will happen but after 2^64 /
+ * 48000 secs, which is pretty large!
+ *
+ * In caln below:
+ *	base may overflow, but since there isn’t any additional division
+ *	performed on base it’s OK
+ *	rem can’t overflow because both are 32-bit values
+ */
+
+#ifdef CONFIG_X86
+static u64 azx_scale64(u64 base, u32 num, u32 den)
+{
+	u64 rem;
+
+	rem = do_div(base, den);
+
+	base *= num;
+	rem *= num;
+
+	do_div(rem, den);
+
+	return base + rem;
+}
+
+static int azx_get_sync_time(ktime_t *device,
+		struct system_counterval_t *system, void *ctx)
+{
+	struct snd_pcm_substream *substream = ctx;
+	struct azx_dev *azx_dev = get_azx_dev(substream);
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct azx *chip = apcm->chip;
+	struct snd_pcm_runtime *runtime;
+	u64 ll_counter, ll_counter_l, ll_counter_h;
+	u64 tsc_counter, tsc_counter_l, tsc_counter_h;
+	u32 wallclk_ctr, wallclk_cycles;
+	bool direction;
+	u32 dma_select;
+	u32 timeout = 200;
+	u32 retry_count = 0;
+
+	runtime = substream->runtime;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		direction = 1;
+	else
+		direction = 0;
+
+	/* 0th stream tag is not used, so DMA ch 0 is for 1st stream tag */
+	do {
+		timeout = 100;
+		dma_select = (direction << GTSCC_CDMAS_DMA_DIR_SHIFT) |
+					(azx_dev->core.stream_tag - 1);
+		snd_hdac_chip_writel(azx_bus(chip), GTSCC, dma_select);
+
+		/* Enable the capture */
+		snd_hdac_chip_updatel(azx_bus(chip), GTSCC, 0, GTSCC_TSCCI_MASK);
+
+		while (timeout) {
+			if (snd_hdac_chip_readl(azx_bus(chip), GTSCC) &
+						GTSCC_TSCCD_MASK)
+				break;
+
+			timeout--;
+		}
+
+		if (!timeout) {
+			dev_err(chip->card->dev, "GTSCC capture Timedout!\n");
+			return -EIO;
+		}
+
+		/* Read wall clock counter */
+		wallclk_ctr = snd_hdac_chip_readl(azx_bus(chip), WALFCC);
+
+		/* Read TSC counter */
+		tsc_counter_l = snd_hdac_chip_readl(azx_bus(chip), TSCCL);
+		tsc_counter_h = snd_hdac_chip_readl(azx_bus(chip), TSCCU);
+
+		/* Read Link counter */
+		ll_counter_l = snd_hdac_chip_readl(azx_bus(chip), LLPCL);
+		ll_counter_h = snd_hdac_chip_readl(azx_bus(chip), LLPCU);
+
+		/* Ack: registers read done */
+		snd_hdac_chip_writel(azx_bus(chip), GTSCC, GTSCC_TSCCD_SHIFT);
+
+		tsc_counter = (tsc_counter_h << TSCCU_CCU_SHIFT) |
+						tsc_counter_l;
+
+		ll_counter = (ll_counter_h << LLPC_CCU_SHIFT) |	ll_counter_l;
+		wallclk_cycles = wallclk_ctr & WALFCC_CIF_MASK;
+
+		/*
+		 * An error occurs near frame "rollover". The clocks in
+		 * frame value indicates whether this error may have
+		 * occurred. Here we use the value of 10 i.e.,
+		 * HDA_MAX_CYCLE_OFFSET
+		 */
+		if (wallclk_cycles < HDA_MAX_CYCLE_VALUE - HDA_MAX_CYCLE_OFFSET
+					&& wallclk_cycles > HDA_MAX_CYCLE_OFFSET)
+			break;
+
+		/*
+		 * Sleep before we read again, else we may again get
+		 * value near to MAX_CYCLE. Try to sleep for different
+		 * amount of time so we dont hit the same number again
+		 */
+		udelay(retry_count++);
+
+	} while (retry_count != HDA_MAX_CYCLE_READ_RETRY);
+
+	if (retry_count == HDA_MAX_CYCLE_READ_RETRY) {
+		dev_err_ratelimited(chip->card->dev,
+			"Error in WALFCC cycle count\n");
+		return -EIO;
+	}
+
+	*device = ns_to_ktime(azx_scale64(ll_counter,
+				NSEC_PER_SEC, runtime->rate));
+	*device = ktime_add_ns(*device, (wallclk_cycles * NSEC_PER_SEC) /
+			       ((HDA_MAX_CYCLE_VALUE + 1) * runtime->rate));
+
+	*system = convert_art_to_tsc(tsc_counter);
+
+	return 0;
+}
+
+#else
+static int azx_get_sync_time(ktime_t *device,
+		struct system_counterval_t *system, void *ctx)
+{
+	return -ENXIO;
+}
+#endif
+
+static int azx_get_crosststamp(struct snd_pcm_substream *substream,
+			      struct system_device_crosststamp *xtstamp)
+{
+	return get_device_system_crosststamp(azx_get_sync_time,
+					substream, NULL, xtstamp);
+}
+
+static inline bool is_link_time_supported(struct snd_pcm_runtime *runtime,
+				struct snd_pcm_audio_tstamp_config *ts)
+{
+	if (runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME)
+		if (ts->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)
+			return true;
+
+	return false;
+}
+
 static int azx_get_time_info(struct snd_pcm_substream *substream,
 			struct timespec *system_ts, struct timespec *audio_ts,
 			struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
 			struct snd_pcm_audio_tstamp_report *audio_tstamp_report)
 {
 	struct azx_dev *azx_dev = get_azx_dev(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct system_device_crosststamp xtstamp;
+	int ret;
 	u64 nsec;
 
 	if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) &&
@@ -361,8 +528,37 @@ static int azx_get_time_info(struct snd_pcm_substream *substream,
 		audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */
 		audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */
 
-	} else
+	} else if (is_link_time_supported(runtime, audio_tstamp_config)) {
+
+		ret = azx_get_crosststamp(substream, &xtstamp);
+		if (ret)
+			return ret;
+
+		switch (runtime->tstamp_type) {
+		case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC:
+			return -EINVAL;
+
+		case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW:
+			*system_ts = ktime_to_timespec(xtstamp.sys_monoraw);
+			break;
+
+		default:
+			*system_ts = ktime_to_timespec(xtstamp.sys_realtime);
+			break;
+
+		}
+
+		*audio_ts = ktime_to_timespec(xtstamp.device);
+
+		audio_tstamp_report->actual_type =
+			SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED;
+		audio_tstamp_report->accuracy_report = 1;
+		/* 24 MHz WallClock == 42ns resolution */
+		audio_tstamp_report->accuracy = 42;
+
+	} else {
 		audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT;
+	}
 
 	return 0;
 }
-- 
2.8.1



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