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authorLinus Torvalds <torvalds@linux-foundation.org>2015-04-15 15:41:41 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2015-04-15 15:41:41 -0700
commitd0a3997c0c3f9351e24029349dee65dd1d9e8d84 (patch)
tree7a04fe282b0c7b329cd87cdb891f0f3879dc71a6 /include/uapi
parent6d50ff91d9780263160262daeb6adfdda8ddbc6c (diff)
parentd6eb9e3ec78c98324097bab8eea266c3bb0d0ac7 (diff)
Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "There have been major modernization with the standard bus: in ALSA sequencer core and HD-audio. Also, HD-audio receives the regmap support replacing the in-house cache register cache code. These changes shouldn't impact the existing behavior, but rather refactoring. In addition, HD-audio got the code split to a core library part and the "legacy" driver parts. This is a preliminary work for adapting the upcoming ASoC HD-audio driver, and the whole transition is still work in progress, likely finished in 4.1. Along with them, there are many updates in ASoC area as usual, too: lots of cleanups, Intel code shuffling, etc. Here are some highlights: ALSA core: - PCM: the audio timestamp / wallclock enhancement - PCM: fixes in DPCM management - Fixes / cleanups of user-space control element management - Sequencer: modernization using the standard bus HD-audio: - Modernization using the standard bus - Regmap support - Use standard runtime PM for codec power saving - Widget-path based power-saving for IDT, VIA and Realtek codecs - Reorganized sysfs entries for each codec object - More Dell headset support ASoC: - Move of jack registration to the card level - Lots of ASoC cleanups, mainly moving things from the CODEC level to the card level - Support for DAPM routes specified by both the machine driver and DT - Continuing improvements to rcar - pcm512x enhacements - Intel platforms updates - rt5670 updates / fixes - New platforms / devices: some non-DSP Qualcomm platforms, Google's Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC Misc: - ice1724: Improved ESI W192M support - emu10k1: Emu 1010 fixes/enhancement" * tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits) ALSA: hda - set GET bit when adding a vendor verb to the codec regmap ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450 ALSA: hda - Fix another race in runtime PM refcounting ALSA: hda - Expose codec type sysfs ALSA: ctl: fix to handle several elements added by one operation for userspace element ASoC: Intel: fix array_size.cocci warnings ASoC: n810: Automatically disconnect non-connected pins ASoC: n810: Consistently pass the card DAPM context to n810_ext_control() ASoC: davinci-evm: Use card DAPM context to access widgets ASoC: mop500_ab8500: Use card DAPM context to access widgets ASoC: wm1133-ev1: Use card DAPM context to access widgets ASoC: atmel: Improve machine driver compile test coverage ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_* ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate ASoC: rnsd: fix build regression without CONFIG_OF ALSA: emu10k1: add toggles for E-mu 1010 optical ports ALSA: ctl: fill identical information to return value when adding userspace elements ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls ALSA: ctl: confirm to return all identical information in 'activate' event ...
Diffstat (limited to 'include/uapi')
-rw-r--r--include/uapi/sound/asequencer.h1
-rw-r--r--include/uapi/sound/asound.h39
-rw-r--r--include/uapi/sound/compress_offload.h2
-rw-r--r--include/uapi/sound/emu10k1.h3
-rw-r--r--include/uapi/sound/hdspm.h6
5 files changed, 43 insertions, 8 deletions
diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h
index 09c8a00ea503..5a5fa4956ebd 100644
--- a/include/uapi/sound/asequencer.h
+++ b/include/uapi/sound/asequencer.h
@@ -22,6 +22,7 @@
#ifndef _UAPI__SOUND_ASEQUENCER_H
#define _UAPI__SOUND_ASEQUENCER_H
+#include <sound/asound.h>
/** version of the sequencer */
#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 0e88e7a0f0eb..46145a5277fe 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -25,6 +25,9 @@
#include <linux/types.h>
+#ifndef __KERNEL__
+#include <stdlib.h>
+#endif
/*
* protocol version
@@ -140,7 +143,7 @@ struct snd_hwdep_dsp_image {
* *
*****************************************************************************/
-#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12)
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
@@ -267,10 +270,17 @@ typedef int __bitwise snd_pcm_subformat_t;
#define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */
#define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */
#define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */
-#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */
+#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */
+#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */
+#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */
+#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */
+#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */
+
#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */
#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */
+
+
typedef int __bitwise snd_pcm_state_t;
#define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */
#define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */
@@ -408,6 +418,22 @@ struct snd_pcm_channel_info {
unsigned int step; /* samples distance in bits */
};
+enum {
+ /*
+ * first definition for backwards compatibility only,
+ * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else
+ */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0,
+
+ /* timestamp definitions */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */
+ SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED
+};
+
struct snd_pcm_status {
snd_pcm_state_t state; /* stream state */
struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */
@@ -419,9 +445,11 @@ struct snd_pcm_status {
snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */
snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */
snd_pcm_state_t suspended_state; /* suspended stream state */
- __u32 reserved_alignment; /* must be filled with zero */
- struct timespec audio_tstamp; /* from sample counter or wall clock */
- unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */
+ __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */
+ struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */
+ struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */
+ __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */
+ unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */
};
struct snd_pcm_mmap_status {
@@ -534,6 +562,7 @@ enum {
#define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t)
#define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22)
#define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr)
+#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status)
#define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info)
#define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40)
#define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41)
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 22ed8cb7800b..e00d8cbfc628 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -75,7 +75,7 @@ struct snd_compr_tstamp {
/**
* struct snd_compr_avail - avail descriptor
* @avail: Number of bytes available in ring buffer for writing/reading
- * @tstamp: timestamp infomation
+ * @tstamp: timestamp information
*/
struct snd_compr_avail {
__u64 avail;
diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h
index d1bbaf78457a..ec1535bb6aed 100644
--- a/include/uapi/sound/emu10k1.h
+++ b/include/uapi/sound/emu10k1.h
@@ -23,8 +23,7 @@
#define _UAPI__SOUND_EMU10K1_H
#include <linux/types.h>
-
-
+#include <sound/asound.h>
/*
* ---- FX8010 ----
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index b357f1a5e29c..5737332d38f2 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -20,6 +20,12 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
+#ifdef __KERNEL__
+#include <linux/types.h>
+#else
+#include <stdint.h>
+#endif
+
/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
#define HDSPM_MAX_CHANNELS 64