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authorLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
committerLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
commit1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch)
tree0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/core/oss/rate.c
downloadblackbird-op-linux-1da177e4c3f41524e886b7f1b8a0c1fc7321cac2.tar.gz
blackbird-op-linux-1da177e4c3f41524e886b7f1b8a0c1fc7321cac2.zip
Linux-2.6.12-rc2v2.6.12-rc2
Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
Diffstat (limited to 'sound/core/oss/rate.c')
-rw-r--r--sound/core/oss/rate.c378
1 files changed, 378 insertions, 0 deletions
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
new file mode 100644
index 000000000000..1096ec186714
--- /dev/null
+++ b/sound/core/oss/rate.c
@@ -0,0 +1,378 @@
+/*
+ * Rate conversion Plug-In
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@suse.cz>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/driver.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+#define SHIFT 11
+#define BITS (1<<SHIFT)
+#define R_MASK (BITS-1)
+
+/*
+ * Basic rate conversion plugin
+ */
+
+typedef struct {
+ signed short last_S1;
+ signed short last_S2;
+} rate_channel_t;
+
+typedef void (*rate_f)(snd_pcm_plugin_t *plugin,
+ const snd_pcm_plugin_channel_t *src_channels,
+ snd_pcm_plugin_channel_t *dst_channels,
+ int src_frames, int dst_frames);
+
+typedef struct rate_private_data {
+ unsigned int pitch;
+ unsigned int pos;
+ rate_f func;
+ int get, put;
+ snd_pcm_sframes_t old_src_frames, old_dst_frames;
+ rate_channel_t channels[0];
+} rate_t;
+
+static void rate_init(snd_pcm_plugin_t *plugin)
+{
+ unsigned int channel;
+ rate_t *data = (rate_t *)plugin->extra_data;
+ data->pos = 0;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ data->channels[channel].last_S1 = 0;
+ data->channels[channel].last_S2 = 0;
+ }
+}
+
+static void resample_expand(snd_pcm_plugin_t *plugin,
+ const snd_pcm_plugin_channel_t *src_channels,
+ snd_pcm_plugin_channel_t *dst_channels,
+ int src_frames, int dst_frames)
+{
+ unsigned int pos = 0;
+ signed int val;
+ signed short S1, S2;
+ char *src, *dst;
+ unsigned int channel;
+ int src_step, dst_step;
+ int src_frames1, dst_frames1;
+ rate_t *data = (rate_t *)plugin->extra_data;
+ rate_channel_t *rchannels = data->channels;
+
+#define GET_S16_LABELS
+#define PUT_S16_LABELS
+#include "plugin_ops.h"
+#undef GET_S16_LABELS
+#undef PUT_S16_LABELS
+ void *get = get_s16_labels[data->get];
+ void *put = put_s16_labels[data->put];
+ signed short sample = 0;
+
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ pos = data->pos;
+ S1 = rchannels->last_S1;
+ S2 = rchannels->last_S2;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, dst_frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = (char *)src_channels[channel].area.addr + src_channels[channel].area.first / 8;
+ dst = (char *)dst_channels[channel].area.addr + dst_channels[channel].area.first / 8;
+ src_step = src_channels[channel].area.step / 8;
+ dst_step = dst_channels[channel].area.step / 8;
+ src_frames1 = src_frames;
+ dst_frames1 = dst_frames;
+ while (dst_frames1-- > 0) {
+ if (pos & ~R_MASK) {
+ pos &= R_MASK;
+ S1 = S2;
+ if (src_frames1-- > 0) {
+ goto *get;
+#define GET_S16_END after_get
+#include "plugin_ops.h"
+#undef GET_S16_END
+ after_get:
+ S2 = sample;
+ src += src_step;
+ }
+ }
+ val = S1 + ((S2 - S1) * (signed int)pos) / BITS;
+ if (val < -32768)
+ val = -32768;
+ else if (val > 32767)
+ val = 32767;
+ sample = val;
+ goto *put;
+#define PUT_S16_END after_put
+#include "plugin_ops.h"
+#undef PUT_S16_END
+ after_put:
+ dst += dst_step;
+ pos += data->pitch;
+ }
+ rchannels->last_S1 = S1;
+ rchannels->last_S2 = S2;
+ rchannels++;
+ }
+ data->pos = pos;
+}
+
+static void resample_shrink(snd_pcm_plugin_t *plugin,
+ const snd_pcm_plugin_channel_t *src_channels,
+ snd_pcm_plugin_channel_t *dst_channels,
+ int src_frames, int dst_frames)
+{
+ unsigned int pos = 0;
+ signed int val;
+ signed short S1, S2;
+ char *src, *dst;
+ unsigned int channel;
+ int src_step, dst_step;
+ int src_frames1, dst_frames1;
+ rate_t *data = (rate_t *)plugin->extra_data;
+ rate_channel_t *rchannels = data->channels;
+
+#define GET_S16_LABELS
+#define PUT_S16_LABELS
+#include "plugin_ops.h"
+#undef GET_S16_LABELS
+#undef PUT_S16_LABELS
+ void *get = get_s16_labels[data->get];
+ void *put = put_s16_labels[data->put];
+ signed short sample = 0;
+
+ for (channel = 0; channel < plugin->src_format.channels; ++channel) {
+ pos = data->pos;
+ S1 = rchannels->last_S1;
+ S2 = rchannels->last_S2;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, dst_frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = (char *)src_channels[channel].area.addr + src_channels[channel].area.first / 8;
+ dst = (char *)dst_channels[channel].area.addr + dst_channels[channel].area.first / 8;
+ src_step = src_channels[channel].area.step / 8;
+ dst_step = dst_channels[channel].area.step / 8;
+ src_frames1 = src_frames;
+ dst_frames1 = dst_frames;
+ while (dst_frames1 > 0) {
+ S1 = S2;
+ if (src_frames1-- > 0) {
+ goto *get;
+#define GET_S16_END after_get
+#include "plugin_ops.h"
+#undef GET_S16_END
+ after_get:
+ S2 = sample;
+ src += src_step;
+ }
+ if (pos & ~R_MASK) {
+ pos &= R_MASK;
+ val = S1 + ((S2 - S1) * (signed int)pos) / BITS;
+ if (val < -32768)
+ val = -32768;
+ else if (val > 32767)
+ val = 32767;
+ sample = val;
+ goto *put;
+#define PUT_S16_END after_put
+#include "plugin_ops.h"
+#undef PUT_S16_END
+ after_put:
+ dst += dst_step;
+ dst_frames1--;
+ }
+ pos += data->pitch;
+ }
+ rchannels->last_S1 = S1;
+ rchannels->last_S2 = S2;
+ rchannels++;
+ }
+ data->pos = pos;
+}
+
+static snd_pcm_sframes_t rate_src_frames(snd_pcm_plugin_t *plugin, snd_pcm_uframes_t frames)
+{
+ rate_t *data;
+ snd_pcm_sframes_t res;
+
+ snd_assert(plugin != NULL, return -ENXIO);
+ if (frames == 0)
+ return 0;
+ data = (rate_t *)plugin->extra_data;
+ if (plugin->src_format.rate < plugin->dst_format.rate) {
+ res = (((frames * data->pitch) + (BITS/2)) >> SHIFT);
+ } else {
+ res = (((frames << SHIFT) + (data->pitch / 2)) / data->pitch);
+ }
+ if (data->old_src_frames > 0) {
+ snd_pcm_sframes_t frames1 = frames, res1 = data->old_dst_frames;
+ while (data->old_src_frames < frames1) {
+ frames1 >>= 1;
+ res1 <<= 1;
+ }
+ while (data->old_src_frames > frames1) {
+ frames1 <<= 1;
+ res1 >>= 1;
+ }
+ if (data->old_src_frames == frames1)
+ return res1;
+ }
+ data->old_src_frames = frames;
+ data->old_dst_frames = res;
+ return res;
+}
+
+static snd_pcm_sframes_t rate_dst_frames(snd_pcm_plugin_t *plugin, snd_pcm_uframes_t frames)
+{
+ rate_t *data;
+ snd_pcm_sframes_t res;
+
+ snd_assert(plugin != NULL, return -ENXIO);
+ if (frames == 0)
+ return 0;
+ data = (rate_t *)plugin->extra_data;
+ if (plugin->src_format.rate < plugin->dst_format.rate) {
+ res = (((frames << SHIFT) + (data->pitch / 2)) / data->pitch);
+ } else {
+ res = (((frames * data->pitch) + (BITS/2)) >> SHIFT);
+ }
+ if (data->old_dst_frames > 0) {
+ snd_pcm_sframes_t frames1 = frames, res1 = data->old_src_frames;
+ while (data->old_dst_frames < frames1) {
+ frames1 >>= 1;
+ res1 <<= 1;
+ }
+ while (data->old_dst_frames > frames1) {
+ frames1 <<= 1;
+ res1 >>= 1;
+ }
+ if (data->old_dst_frames == frames1)
+ return res1;
+ }
+ data->old_dst_frames = frames;
+ data->old_src_frames = res;
+ return res;
+}
+
+static snd_pcm_sframes_t rate_transfer(snd_pcm_plugin_t *plugin,
+ const snd_pcm_plugin_channel_t *src_channels,
+ snd_pcm_plugin_channel_t *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ snd_pcm_uframes_t dst_frames;
+ rate_t *data;
+
+ snd_assert(plugin != NULL && src_channels != NULL && dst_channels != NULL, return -ENXIO);
+ if (frames == 0)
+ return 0;
+#ifdef CONFIG_SND_DEBUG
+ {
+ unsigned int channel;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ snd_assert(src_channels[channel].area.first % 8 == 0 &&
+ src_channels[channel].area.step % 8 == 0,
+ return -ENXIO);
+ snd_assert(dst_channels[channel].area.first % 8 == 0 &&
+ dst_channels[channel].area.step % 8 == 0,
+ return -ENXIO);
+ }
+ }
+#endif
+
+ dst_frames = rate_dst_frames(plugin, frames);
+ if (dst_frames > dst_channels[0].frames)
+ dst_frames = dst_channels[0].frames;
+ data = (rate_t *)plugin->extra_data;
+ data->func(plugin, src_channels, dst_channels, frames, dst_frames);
+ return dst_frames;
+}
+
+static int rate_action(snd_pcm_plugin_t *plugin,
+ snd_pcm_plugin_action_t action,
+ unsigned long udata ATTRIBUTE_UNUSED)
+{
+ snd_assert(plugin != NULL, return -ENXIO);
+ switch (action) {
+ case INIT:
+ case PREPARE:
+ rate_init(plugin);
+ break;
+ default:
+ break;
+ }
+ return 0; /* silenty ignore other actions */
+}
+
+int snd_pcm_plugin_build_rate(snd_pcm_plug_t *plug,
+ snd_pcm_plugin_format_t *src_format,
+ snd_pcm_plugin_format_t *dst_format,
+ snd_pcm_plugin_t **r_plugin)
+{
+ int err;
+ rate_t *data;
+ snd_pcm_plugin_t *plugin;
+
+ snd_assert(r_plugin != NULL, return -ENXIO);
+ *r_plugin = NULL;
+
+ snd_assert(src_format->channels == dst_format->channels, return -ENXIO);
+ snd_assert(src_format->channels > 0, return -ENXIO);
+ snd_assert(snd_pcm_format_linear(src_format->format) != 0, return -ENXIO);
+ snd_assert(snd_pcm_format_linear(dst_format->format) != 0, return -ENXIO);
+ snd_assert(src_format->rate != dst_format->rate, return -ENXIO);
+
+ err = snd_pcm_plugin_build(plug, "rate conversion",
+ src_format, dst_format,
+ sizeof(rate_t) + src_format->channels * sizeof(rate_channel_t),
+ &plugin);
+ if (err < 0)
+ return err;
+ data = (rate_t *)plugin->extra_data;
+ data->get = getput_index(src_format->format);
+ snd_assert(data->get >= 0 && data->get < 4*2*2, return -EINVAL);
+ data->put = getput_index(dst_format->format);
+ snd_assert(data->put >= 0 && data->put < 4*2*2, return -EINVAL);
+
+ if (src_format->rate < dst_format->rate) {
+ data->pitch = ((src_format->rate << SHIFT) + (dst_format->rate >> 1)) / dst_format->rate;
+ data->func = resample_expand;
+ } else {
+ data->pitch = ((dst_format->rate << SHIFT) + (src_format->rate >> 1)) / src_format->rate;
+ data->func = resample_shrink;
+ }
+ data->pos = 0;
+ rate_init(plugin);
+ data->old_src_frames = data->old_dst_frames = 0;
+ plugin->transfer = rate_transfer;
+ plugin->src_frames = rate_src_frames;
+ plugin->dst_frames = rate_dst_frames;
+ plugin->action = rate_action;
+ *r_plugin = plugin;
+ return 0;
+}
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