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authorAriff Abdullah <ariff@FreeBSD.org>2009-06-07 19:12:08 +0000
committerAriff Abdullah <ariff@FreeBSD.org>2009-06-07 19:12:08 +0000
commit90da2b2859b259fca1ab223931f549a72e21a3a5 (patch)
tree906f402638735c3f32e226f6868f207db569d6a9 /sys/dev/sound/pci/via8233.c
parent0a276edef96edc86d1af97b39f76ad168599ceb4 (diff)
downloadsrc-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/pci/via8233.c')
-rw-r--r--sys/dev/sound/pci/via8233.c51
1 files changed, 25 insertions, 26 deletions
diff --git a/sys/dev/sound/pci/via8233.c b/sys/dev/sound/pci/via8233.c
index 68de81b7e5f8..1f5f48299f20 100644
--- a/sys/dev/sound/pci/via8233.c
+++ b/sys/dev/sound/pci/via8233.c
@@ -35,6 +35,10 @@
* ordering.
*/
+#ifdef HAVE_KERNEL_OPTION_HEADERS
+#include "opt_snd.h"
+#endif
+
#include <dev/sound/pcm/sound.h>
#include <dev/sound/pcm/ac97.h>
@@ -123,10 +127,10 @@ struct via_info {
};
static uint32_t via_fmt[] = {
- AFMT_U8,
- AFMT_STEREO | AFMT_U8,
- AFMT_S16_LE,
- AFMT_STEREO | AFMT_S16_LE,
+ SND_FORMAT(AFMT_U8, 1, 0),
+ SND_FORMAT(AFMT_U8, 2, 0),
+ SND_FORMAT(AFMT_S16_LE, 1, 0),
+ SND_FORMAT(AFMT_S16_LE, 2, 0),
0
};
@@ -150,7 +154,6 @@ via_chan_active(struct via_info *via)
return (ret);
}
-#ifdef SND_DYNSYSCTL
static int
sysctl_via8233_spdif_enable(SYSCTL_HANDLER_ARGS)
{
@@ -243,12 +246,10 @@ sysctl_via_polling(SYSCTL_HANDLER_ARGS)
return (err);
}
-#endif /* SND_DYNSYSCTL */
static void
via_init_sysctls(device_t dev)
{
-#ifdef SND_DYNSYSCTL
/* XXX: an user should be able to set this with a control tool,
if not done before 7.0-RELEASE, this needs to be converted to
a device specific sysctl "dev.pcm.X.yyy" via device_get_sysctl_*()
@@ -268,7 +269,6 @@ via_init_sysctls(device_t dev)
"polling", CTLTYPE_INT | CTLFLAG_RW, dev, sizeof(dev),
sysctl_via_polling, "I",
"Enable polling mode");
-#endif
}
static __inline uint32_t
@@ -374,7 +374,7 @@ via_read_codec(kobj_t obj, void *addr, int reg)
static kobj_method_t via_ac97_methods[] = {
KOBJMETHOD(ac97_read, via_read_codec),
KOBJMETHOD(ac97_write, via_write_codec),
- { 0, 0 }
+ KOBJMETHOD_END
};
AC97_DECLARE(via_ac97);
@@ -408,7 +408,7 @@ via8233wr_setformat(kobj_t obj, void *data, uint32_t format)
uint32_t f = WR_FORMAT_STOP_INDEX;
- if (format & AFMT_STEREO)
+ if (AFMT_CHANNEL(format) > 1)
f |= WR_FORMAT_STEREO;
if (format & AFMT_S16_LE)
f |= WR_FORMAT_16BIT;
@@ -431,7 +431,7 @@ via8233dxs_setformat(kobj_t obj, void *data, uint32_t format)
v = via_rd(via, r, 4);
v &= ~(VIA8233_DXS_RATEFMT_STEREO | VIA8233_DXS_RATEFMT_16BIT);
- if (format & AFMT_STEREO)
+ if (AFMT_CHANNEL(format) > 1)
v |= VIA8233_DXS_RATEFMT_STEREO;
if (format & AFMT_16BIT)
v |= VIA8233_DXS_RATEFMT_16BIT;
@@ -450,7 +450,7 @@ via8233msgd_setformat(kobj_t obj, void *data, uint32_t format)
uint32_t s = 0xff000000;
uint8_t v = (format & AFMT_S16_LE) ? MC_SGD_16BIT : MC_SGD_8BIT;
- if (format & AFMT_STEREO) {
+ if (AFMT_CHANNEL(format) > 1) {
v |= MC_SGD_CHANNELS(2);
s |= SLOT3(1) | SLOT4(2);
} else {
@@ -469,7 +469,7 @@ via8233msgd_setformat(kobj_t obj, void *data, uint32_t format)
/* -------------------------------------------------------------------- */
/* Speed setting functions */
-static int
+static uint32_t
via8233wr_setspeed(kobj_t obj, void *data, uint32_t speed)
{
struct via_chinfo *ch = data;
@@ -481,7 +481,7 @@ via8233wr_setspeed(kobj_t obj, void *data, uint32_t speed)
return (48000);
}
-static int
+static uint32_t
via8233dxs_setspeed(kobj_t obj, void *data, uint32_t speed)
{
struct via_chinfo *ch = data;
@@ -501,7 +501,7 @@ via8233dxs_setspeed(kobj_t obj, void *data, uint32_t speed)
return (speed);
}
-static int
+static uint32_t
via8233msgd_setspeed(kobj_t obj, void *data, uint32_t speed)
{
struct via_chinfo *ch = data;
@@ -596,10 +596,10 @@ via8233chan_setfragments(kobj_t obj, void *data,
ch->blksz = sndbuf_getblksz(ch->buffer);
ch->blkcnt = sndbuf_getblkcnt(ch->buffer);
- return (1);
+ return (0);
}
-static int
+static uint32_t
via8233chan_setblocksize(kobj_t obj, void *data, uint32_t blksz)
{
struct via_chinfo *ch = data;
@@ -610,13 +610,12 @@ via8233chan_setblocksize(kobj_t obj, void *data, uint32_t blksz)
return (ch->blksz);
}
-static int
+static uint32_t
via8233chan_getptr(kobj_t obj, void *data)
{
struct via_chinfo *ch = data;
struct via_info *via = ch->parent;
- uint32_t v, index, count;
- int ptr;
+ uint32_t v, index, count, ptr;
snd_mtxlock(via->lock);
if (via->polling != 0) {
@@ -852,7 +851,7 @@ via_poll_ticks(struct via_info *via)
if (ch->channel == NULL || ch->active == 0)
continue;
pollticks = ((uint64_t)hz * ch->blksz) /
- ((uint64_t)sndbuf_getbps(ch->buffer) *
+ ((uint64_t)sndbuf_getalign(ch->buffer) *
sndbuf_getspd(ch->buffer));
pollticks >>= 2;
if (pollticks > hz)
@@ -868,7 +867,7 @@ via_poll_ticks(struct via_info *via)
if (ch->channel == NULL || ch->active == 0)
continue;
pollticks = ((uint64_t)hz * ch->blksz) /
- ((uint64_t)sndbuf_getbps(ch->buffer) *
+ ((uint64_t)sndbuf_getalign(ch->buffer) *
sndbuf_getspd(ch->buffer));
pollticks >>= 2;
if (pollticks > hz)
@@ -902,7 +901,7 @@ via8233chan_trigger(kobj_t obj, void* data, int go)
ch->ptr = 0;
ch->prevptr = 0;
pollticks = ((uint64_t)hz * ch->blksz) /
- ((uint64_t)sndbuf_getbps(ch->buffer) *
+ ((uint64_t)sndbuf_getalign(ch->buffer) *
sndbuf_getspd(ch->buffer));
pollticks >>= 2;
if (pollticks > hz)
@@ -974,7 +973,7 @@ static kobj_method_t via8233wr_methods[] = {
KOBJMETHOD(channel_setfragments, via8233chan_setfragments),
KOBJMETHOD(channel_trigger, via8233chan_trigger),
KOBJMETHOD(channel_getptr, via8233chan_getptr),
- { 0, 0 }
+ KOBJMETHOD_END
};
CHANNEL_DECLARE(via8233wr);
@@ -987,7 +986,7 @@ static kobj_method_t via8233dxs_methods[] = {
KOBJMETHOD(channel_setfragments, via8233chan_setfragments),
KOBJMETHOD(channel_trigger, via8233chan_trigger),
KOBJMETHOD(channel_getptr, via8233chan_getptr),
- { 0, 0 }
+ KOBJMETHOD_END
};
CHANNEL_DECLARE(via8233dxs);
@@ -1000,7 +999,7 @@ static kobj_method_t via8233msgd_methods[] = {
KOBJMETHOD(channel_setfragments, via8233chan_setfragments),
KOBJMETHOD(channel_trigger, via8233chan_trigger),
KOBJMETHOD(channel_getptr, via8233chan_getptr),
- { 0, 0 }
+ KOBJMETHOD_END
};
CHANNEL_DECLARE(via8233msgd);