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author | Ariff Abdullah <ariff@FreeBSD.org> | 2009-06-07 19:12:08 +0000 |
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committer | Ariff Abdullah <ariff@FreeBSD.org> | 2009-06-07 19:12:08 +0000 |
commit | 90da2b2859b259fca1ab223931f549a72e21a3a5 (patch) | |
tree | 906f402638735c3f32e226f6868f207db569d6a9 /sys/dev/sound/sbus/cs4231.c | |
parent | 0a276edef96edc86d1af97b39f76ad168599ceb4 (diff) | |
download | src-90da2b2859b259fca1ab223931f549a72e21a3a5.tar.gz src-90da2b2859b259fca1ab223931f549a72e21a3a5.zip |
Sound Mega-commit. Expect further cleanup until code freeze.
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 <avg@freebsd.org>
* 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 <olivier at gid0 d org>, too many
unsung / unnamed heroes.
Notes
Notes:
svn path=/head/; revision=193640
Diffstat (limited to 'sys/dev/sound/sbus/cs4231.c')
-rw-r--r-- | sys/dev/sound/sbus/cs4231.c | 36 |
1 files changed, 20 insertions, 16 deletions
diff --git a/sys/dev/sound/sbus/cs4231.c b/sys/dev/sound/sbus/cs4231.c index a785f96dbe56..90c947f5aaf3 100644 --- a/sys/dev/sound/sbus/cs4231.c +++ b/sys/dev/sound/sbus/cs4231.c @@ -50,6 +50,10 @@ __FBSDID("$FreeBSD$"); #include <machine/bus.h> #include <machine/ofw_machdep.h> +#ifdef HAVE_KERNEL_OPTION_HEADERS +#include "opt_snd.h" +#endif + #include <dev/sound/pcm/sound.h> #include <dev/sound/sbus/apcdmareg.h> #include <dev/sound/sbus/cs4231.h> @@ -260,18 +264,18 @@ MODULE_VERSION(snd_audiocs, 1); static u_int32_t cs4231_fmt[] = { - AFMT_U8, - AFMT_STEREO | AFMT_U8, - AFMT_MU_LAW, - AFMT_STEREO | AFMT_MU_LAW, - AFMT_A_LAW, - AFMT_STEREO | AFMT_A_LAW, - AFMT_IMA_ADPCM, - AFMT_STEREO | AFMT_IMA_ADPCM, - AFMT_S16_LE, - AFMT_STEREO | AFMT_S16_LE, - AFMT_S16_BE, - AFMT_STEREO | AFMT_S16_BE, + SND_FORMAT(AFMT_U8, 1, 0), + SND_FORMAT(AFMT_U8, 2, 0), + SND_FORMAT(AFMT_MU_LAW, 1, 0), + SND_FORMAT(AFMT_MU_LAW, 2, 0), + SND_FORMAT(AFMT_A_LAW, 1, 0), + SND_FORMAT(AFMT_A_LAW, 2, 0), + SND_FORMAT(AFMT_IMA_ADPCM, 1, 0), + SND_FORMAT(AFMT_IMA_ADPCM, 2, 0), + SND_FORMAT(AFMT_S16_LE, 1, 0), + SND_FORMAT(S16_LE, 2, 0), + SND_FORMAT(AFMT_S16_BE, 1, 0), + SND_FORMAT(AFMT_S16_BE, 2, 0), 0 }; @@ -288,7 +292,7 @@ static kobj_method_t cs4231_chan_methods[] = { KOBJMETHOD(channel_trigger, cs4231_chan_trigger), KOBJMETHOD(channel_getptr, cs4231_chan_getptr), KOBJMETHOD(channel_getcaps, cs4231_chan_getcaps), - { 0, 0 } + KOBJMETHOD_END }; CHANNEL_DECLARE(cs4231_chan); @@ -299,7 +303,7 @@ static kobj_method_t cs4231_mixer_methods[] = { KOBJMETHOD(mixer_init, cs4231_mixer_init), KOBJMETHOD(mixer_set, cs4231_mixer_set), KOBJMETHOD(mixer_setrecsrc, cs4231_mixer_setrecsrc), - { 0, 0 } + KOBJMETHOD_END }; MIXER_DECLARE(cs4231_mixer); @@ -1057,7 +1061,7 @@ cs4231_chan_setformat(kobj_t obj, void *data, u_int32_t format) return (0); } - encoding = format & ~AFMT_STEREO; + encoding = AFMT_ENCODING(format); fs = 0; switch (encoding) { case AFMT_U8: @@ -1084,7 +1088,7 @@ cs4231_chan_setformat(kobj_t obj, void *data, u_int32_t format) break; } - if (format & AFMT_STEREO) + if (AFMT_CHANNEL(format) > 1) fs |= CS_AFMT_STEREO; DPRINTF(("FORMAT: %s : 0x%x\n", ch->dir == PCMDIR_PLAY ? "playback" : |