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author | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 15:20:36 -0700 |
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committer | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 15:20:36 -0700 |
commit | 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch) | |
tree | 0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/core/oss/rate.c | |
download | blackbird-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.c | 378 |
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; +} |