[alsa-devel] [PATCH 2/3] ALSA - hda: Add support for link audio time reporting
Takashi Iwai
tiwai at suse.de
Mon Jul 11 12:32:07 CEST 2016
On Mon, 11 Jul 2016 12:13:28 +0200,
Vinod Koul wrote:
>
> From: Guneshwor Singh <guneshwor.o.singh at intel.com>
>
> Skylake onwards HDA controller supports reprting link audio
> time, so add support for that.
It's way too few description, the text is almost same as the previous
patch. Please give more information.
>
> 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: Vinod Koul <vinod.koul at intel.com>
> ---
> sound/pci/hda/hda_controller.c | 159 ++++++++++++++++++++++++++++++++++++++++-
> 1 file changed, 158 insertions(+), 1 deletion(-)
>
> diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
> index 9833666c6108..5613a403d720 100644
> --- a/sound/pci/hda/hda_controller.c
> +++ b/sound/pci/hda/hda_controller.c
> @@ -27,6 +27,7 @@
> #include <linux/module.h>
> #include <linux/pm_runtime.h>
> #include <linux/slab.h>
> +#include <asm/tsc.h>
> #include <sound/core.h>
> #include <sound/initval.h>
> #include "hda_controller.h"
> @@ -34,6 +35,8 @@
> #define CREATE_TRACE_POINTS
> #include "hda_controller_trace.h"
>
> +#define SEC_TO_NSEC 1000000000LL
Can we use a definition in time64.h?
> /* DSP lock helpers */
> #define dsp_lock(dev) snd_hdac_dsp_lock(azx_stream(dev))
> #define dsp_unlock(dev) snd_hdac_dsp_unlock(azx_stream(dev))
> @@ -337,12 +340,136 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
> azx_get_position(chip, azx_dev));
> }
>
> +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;
> +}
What is this function supposed to do?
> +static int azx_get_sync_time(ktime_t *device,
> + struct system_counterval_t *system, void *ctx)
> +{
> + struct snd_pcm_substream *substream = (struct snd_pcm_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_write(l, azx_bus(chip), AZX_REG_GTSCC,
> + dma_select);
> + /* Enable the capture */
> + _snd_hdac_chip_write(l, azx_bus(chip), AZX_REG_GTSCC,
> + _snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_GTSCC) | GTSCC_TSCCI_MASK);
> +
> + while (timeout) {
> + if (_snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_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_read(l, azx_bus(chip),
> + AZX_REG_WALFCC);
> +
> + /* Read TSC counter */
> + tsc_counter_l = _snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_TSCCL);
> + tsc_counter_h = _snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_TSCCU);
> +
> + /* Read Link counter */
> + ll_counter_l = _snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_LLPCL);
> + ll_counter_h = _snd_hdac_chip_read(l, azx_bus(chip),
> + AZX_REG_LLPCU);
> +
> + /* Ack: registers read done */
> + _snd_hdac_chip_write(l, azx_bus(chip),
> + AZX_REG_GTSCC,
> + (0x1 << 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;
> +
> + if (wallclk_cycles < HDA_MAX_CYCLE_VALUE - HDA_MAX_CYCLE_OFFSET
> + && wallclk_cycles > HDA_MAX_CYCLE_OFFSET)
> + break;
Is this condition really correct...? It's hard to understand.
> +
> + /*
> + * 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(chip->card->dev, "Error in WALFCC cycle count\n");
> + return -EIO;
> + }
> +
> + *device = ns_to_ktime(azx_scale64(ll_counter,
> + SEC_TO_NSEC, runtime->rate));
> + *device = ktime_add_ns(*device, (wallclk_cycles * SEC_TO_NSEC) /
> + ((HDA_MAX_CYCLE_VALUE+1) * runtime->rate));
Hmm, the calculation here looks as if there can be an optimization...
thanks,
Takashi
> + *system = convert_art_to_tsc(tsc_counter);
> +
> + return 0;
> +}
> +
> +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 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;
> u64 nsec;
>
> if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) &&
> @@ -361,8 +488,38 @@ 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 ((runtime->hw.info &
> + SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME) &&
> + (audio_tstamp_config->type_requested ==
> + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)) {
> +
> + azx_get_crosststamp(substream, &xtstamp);
> +
> + 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;
> }
> --
> 1.9.1
>
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