For allowing adjusting the timestamp type on the fly, add it to
sw_params. The existing ioctl is still kept for compatibility.
Along with this, increment the PCM protocol version.
The extension was suggested by Clemens Ladisch.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For applications which need to synchronise with external timebases such
as broadcast TV applications the kernel monotonic time is not optimal as
it includes adjustments from NTP and so may still include discontinuities
due to that. A raw monotonic time which does not include any adjustments
is available in the kernel from getrawmonotonic() so provide userspace with
a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides
timestamps based on this as an option.
[dropped tstamp_type assignment code, as it's no longer needed -- tiwai]
Reported-by: Daniel Thompson <daniel.thompson@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. To use hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a start point for further development, this is an incomplete driver
for DICE devices:
- only playback (so no clock source except the bus clock)
- only 44.1 kHz
- no MIDI
- recovery after bus reset is slow
- hwdep device is created, but not actually implemented
Contains compilation fixes by Stefan Richter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The char arrays with size 44 are for the name string of
snd_ctl_elem_id. Define the constant and replace the raw numbers with
it for clarifying better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 4eeaaeaea (ALSA: core: add hooks for audio timestamps) added the
new audio_tstamp field to struct snd_pcm_status. However, struct
timespec requires 64-bit alignment, so the 64-bit compiler would insert
32 bits of padding before this field, which broke SNDRV_PCM_IOCTL_STATUS
with error messages like this:
kernel: unknown ioctl = 0x80984120
To solve this, insert the padding explicitly so that it can be taken
into account when calculating the ABI structure size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
