src/sys/dev/sound/pcm/buffer.h, branch upstream/11.2.0

Revert r330897:

2018-03-29T02:50:57Z

This was intended to be a non-functional change. It wasn't. The commit message was thus wrong. In addition it broke arm, and merged crypto related code. Revert with prejudice. This revert skips files touched in r316370 since that commit was since MFCed. This revert also skips files that require $FreeBSD$ property changes. Thank you to those who helped me get out of this mess including but not limited to gonzo, kevans, rgrimes. Requested by: gjb (re)

Partial merge of the SPDX changes

2018-03-14T03:19:51Z

These changes are incomplete but are making it difficult to determine what other changes can/should be merged. No objections from: pfg

Make sound(4) more flexible in setting soft buffer and block sizes when

2012-01-31T21:46:28Z

hardware imposes strict limitations on hard buffer and block sizes. Previous code set soft buffer to be no smaller then hard buffer. On some cards with fixed 64K physical buffer that caused up to 800ms play latency. New code allows to set soft buffer size down to just two blocks of the hard buffer and to not write more then that size ahead to the hardware buffer. As result of that change I was able to reduce full practically measured record-playback loop delay in those conditions down to only about 115ms with theoretical playback latency of only about 50ms. New code works fine for both vchans and direct cases. In both cases sound(4) tries to follow hw.snd.latency_profile and hw.snd.latency values and application-requested buffer and block sizes as much as limitation of two hardware blocks allows. Reviewed by: silence on multimedia@

- Remove more dead code[1]. Since r207330, we only need to check division

2010-05-04T16:56:59Z

by zero of the second argument 'from'. - Prefer u_int32_t over unsigned int to make its intention more clearer. - Move the function to a header file and make it a static inline function. Pointed out by: Andrew Reilly (areilly at bigpond dot net dot au)[1] MFC after: 3 days

Sound Mega-commit. Expect further cleanup until code freeze.

2009-06-07T19:12:08Z

For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS , too many unsung / unnamed heroes.

Buffer optimization and locking cleanup. Don't resize/malloc

2007-06-14T11:15:51Z

unless it is really necessary to ease down unlock/lock sequence.

buf_addr should be bus_addr_t rather than u_int32_t.

2007-05-07T02:46:48Z

sndbuf_alloc() now accept dmaflags argument which will be forwarded to

2007-04-18T18:26:41Z

internal bus_dmammem_alloc() for greater flexibility on setting up DMA / page attributes.

Fix huge memory leak within sound buffer (during channel destruction,

2007-02-01T09:46:03Z

buffer resizing, etc.) that was here since eon. Free all (unmanaged) allocated buffer through sndbuf_destroy() in case we forgot to call sndbuf_free(). For a managed buffer (mostly hw specific managed buffer), either provide CHANNEL_FREE() method with appropriate return value to invoke semi-automatic sndbuf_free() or simply do it on their own. If everything is failed, sndbuf_destroy() will come to the rescue as a final measure. MFC after: 3 days

Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes

2006-11-26T12:24:06Z

in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.