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* ALSA: compress: add num_sample_rates in snd_codec_descVinod Koul2014-01-071-0/+2
| | | | | | | | this gives ability to convey the valid values of supported rates in sample_rates array Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: compress: update struct snd_codec_desc for sample rateVinod Koul2014-01-051-1/+2
| | | | | | | | | Now that we don't use SNDRV_PCM_RATE_xxx bit fields for sample rate, we need to change the description to an array for describing the sample rates supported by the sink/source Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: compress: update comment for sample rate in snd_codecVinod Koul2014-01-051-1/+2
| | | | | Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: compress: change the way sample rates are sent to kernelVinod Koul2013-12-161-1/+1
| | | | | | | | | The usage of SNDRV_RATES is not effective as we can have rates like 12000 or some other ones used by decoders. This change the usage of this to use the raw Hz values to be sent to kernel Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: compress: Fix 64bit ABI incompatibilityTakashi Iwai2013-12-101-3/+3
| | | | | | | | | | | | | snd_pcm_uframes_t is defined as unsigned long so it would take different sizes depending on 32 or 64bit architectures. As we don't want this ABI incompatibility, and there is no real 64bit user yet, let's make it the fixed size with __u32. Also bump the protocol version number to 0.1.2. Acked-by: Vinod Koul <vinod.koul@intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: include/uapi/sound/firewire.h: use "_UAPI" instead of "UAPI"Chen Gang2013-11-071-3/+3
| | | | | | | | | | | | | When installing, "scripts/headers_install.sh" will strip guard macro' "_UAPI" to prevent from appearing it to users. And also, all another files which need uapi prefix always use "_UAPI", not "UAPI". So use "_UAPI" instead of "UAPI" on the guard macro, and also give a comment for "#endif". Signed-off-by: Chen Gang <gang.chen@asianux.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'dice-driver-playback-only' of ↵Takashi Iwai2013-10-223-1/+54
|\ | | | | | | git://git.alsa-project.org/alsa-kprivate into for-next
| * ALSA: add DICE driverClemens Ladisch2013-10-173-1/+54
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTCAdrian Knoth2013-08-191-1/+1
|/ | | | | | | | | | | Use struct hdspm_ltc to query the LTC, using a mixer struct is just plain wrong. Due to the wrong struct, this ioctl was never working, so we're free to fix it without breaking userspace compatibility. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Replace the magic number 44 with constTakashi Iwai2013-06-281-0/+2
| | | | | | | | 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>
* ALSA: add DSD formatsDaniel Mack2013-04-181-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* ALSA: compress: add support for gapless playbackJeeja KP2013-02-141-1/+30
| | | | | | | | | | | | | this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Extend chmap definitions for UAC2Takashi Iwai2012-11-261-1/+11
| | | | | | | USB audio class 2 has more channel map positions than we currently have. Let's add missing definitions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: core: fix 64-bit SNDRV_PCM_IOCTL_STATUS ABI breakageClemens Ladisch2012-10-281-1/+2
| | | | | | | | | | | | | | | | 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: core: add hooks for audio timestampsPierre-Louis Bossart2012-10-231-2/+5
| | | | | | | | | | | | | | | | | | | 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>
* UAPI: (Scripted) Disintegrate include/soundDavid Howells2012-10-0911-0/+3322
| | | | | | | | | 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>
* UAPI: (Scripted) Set up UAPI Kbuild filesDavid Howells2012-10-021-0/+1
Set up empty UAPI Kbuild files to be populated by the header splitter. Signed-off-by: David Howells <dhowells@redhat.com> Acked-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Thomas Gleixner <tglx@linutronix.de> Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com> Acked-by: Dave Jones <davej@redhat.com>
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