[SCM] libav/experimental: Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder. patch by Kenan Gillet, kenan.gillet gmail com
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 16:31:39 UTC 2013
The following commit has been merged in the experimental branch:
commit 4599d22c0cd7fa952bc02375d2899d8c9a31b9ae
Author: Kenan Gillet <kenan.gillet at gmail.com>
Date: Fri Oct 24 21:29:23 2008 +0000
Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
patch by Kenan Gillet, kenan.gillet gmail com
Originally committed as revision 15680 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index ece4a96..7c9bbd4 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -156,7 +156,7 @@ OBJS-$(CONFIG_QDRAW_DECODER) += qdrw.o
OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o
OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o
OBJS-$(CONFIG_QTRLE_ENCODER) += qtrleenc.o
-OBJS-$(CONFIG_RA_144_DECODER) += ra144.o acelp_filters.o
+OBJS-$(CONFIG_RA_144_DECODER) += ra144.o celp_filters.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c
index 59db649..94ec947 100644
--- a/libavcodec/acelp_filters.c
+++ b/libavcodec/acelp_filters.c
@@ -81,65 +81,6 @@ void ff_acelp_interpolate(
}
}
-void ff_acelp_convolve_circ(
- int16_t* fc_out,
- const int16_t* fc_in,
- const int16_t* filter,
- int len)
-{
- int i, k;
-
- memset(fc_out, 0, len * sizeof(int16_t));
-
- /* Since there are few pulses over an entire subframe (i.e. almost
- all fc_in[i] are zero) it is faster to loop over fc_in first. */
- for(i=0; i<len; i++)
- {
- if(fc_in[i])
- {
- for(k=0; k<i; k++)
- fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;
-
- for(k=i; k<len; k++)
- fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15;
- }
- }
-}
-
-int ff_acelp_lp_synthesis_filter(
- int16_t *out,
- const int16_t* filter_coeffs,
- const int16_t* in,
- int buffer_length,
- int filter_length,
- int stop_on_overflow,
- int rounder)
-{
- int i,n;
-
- // These two lines are to avoid a -1 subtraction in the main loop
- filter_length++;
- filter_coeffs--;
-
- for(n=0; n<buffer_length; n++)
- {
- int sum = rounder;
- for(i=1; i<filter_length; i++)
- sum -= filter_coeffs[i] * out[n-i];
-
- sum = (sum >> 12) + in[n];
-
- if(sum + 0x8000 > 0xFFFFU)
- {
- if(stop_on_overflow)
- return 1;
- sum = (sum >> 31) ^ 32767;
- }
- out[n] = sum;
- }
-
- return 0;
-}
void ff_acelp_high_pass_filter(
int16_t* out,
diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index b2f05bc..e1e3d68 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -60,50 +60,6 @@ void ff_acelp_interpolate(
int filter_length,
int length);
-/**
- * Circularly convolve fixed vector with a phase dispersion impulse
- * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
- * @param fc_out vector with filter applied
- * @param fc_in source vector
- * @param filter phase filter coefficients
- *
- * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
- *
- * \note fc_in and fc_out should not overlap!
- */
-void ff_acelp_convolve_circ(
- int16_t* fc_out,
- const int16_t* fc_in,
- const int16_t* filter,
- int len);
-
-/**
- * LP synthesis filter.
- * @param out [out] pointer to output buffer
- * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
- * @param in input signal
- * @param buffer_length amount of data to process
- * @param filter_length filter length (10 for 10th order LP filter)
- * @param stop_on_overflow 1 - return immediately if overflow occurs
- * 0 - ignore overflows
- * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
- *
- * @return 1 if overflow occurred, 0 - otherwise
- *
- * @note Output buffer must contain 10 samples of past
- * speech data before pointer.
- *
- * Routine applies 1/A(z) filter to given speech data.
- */
-int ff_acelp_lp_synthesis_filter(
- int16_t *out,
- const int16_t* filter_coeffs,
- const int16_t* in,
- int buffer_length,
- int filter_length,
- int stop_on_overflow,
- int rounder);
-
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
diff --git a/libavcodec/celp_filters.c b/libavcodec/celp_filters.c
new file mode 100644
index 0000000..758c9b0
--- /dev/null
+++ b/libavcodec/celp_filters.c
@@ -0,0 +1,86 @@
+/*
+ * various filters for ACELP-based codecs
+ *
+ * Copyright (c) 2008 Vladimir Voroshilov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <inttypes.h>
+
+#include "avcodec.h"
+#include "celp_filters.h"
+
+void ff_celp_convolve_circ(
+ int16_t* fc_out,
+ const int16_t* fc_in,
+ const int16_t* filter,
+ int len)
+{
+ int i, k;
+
+ memset(fc_out, 0, len * sizeof(int16_t));
+
+ /* Since there are few pulses over an entire subframe (i.e. almost
+ all fc_in[i] are zero) it is faster to loop over fc_in first. */
+ for(i=0; i<len; i++)
+ {
+ if(fc_in[i])
+ {
+ for(k=0; k<i; k++)
+ fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;
+
+ for(k=i; k<len; k++)
+ fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15;
+ }
+ }
+}
+
+int ff_celp_lp_synthesis_filter(
+ int16_t *out,
+ const int16_t* filter_coeffs,
+ const int16_t* in,
+ int buffer_length,
+ int filter_length,
+ int stop_on_overflow,
+ int rounder)
+{
+ int i,n;
+
+ // These two lines are to avoid a -1 subtraction in the main loop
+ filter_length++;
+ filter_coeffs--;
+
+ for(n=0; n<buffer_length; n++)
+ {
+ int sum = rounder;
+ for(i=1; i<filter_length; i++)
+ sum -= filter_coeffs[i] * out[n-i];
+
+ sum = (sum >> 12) + in[n];
+
+ if(sum + 0x8000 > 0xFFFFU)
+ {
+ if(stop_on_overflow)
+ return 1;
+ sum = (sum >> 31) ^ 32767;
+ }
+ out[n] = sum;
+ }
+
+ return 0;
+}
diff --git a/libavcodec/celp_filters.h b/libavcodec/celp_filters.h
new file mode 100644
index 0000000..cb73aa8
--- /dev/null
+++ b/libavcodec/celp_filters.h
@@ -0,0 +1,72 @@
+/*
+ * various filters for CELP-based codecs
+ *
+ * Copyright (c) 2008 Vladimir Voroshilov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_CELP_FILTERS_H
+#define AVCODEC_CELP_FILTERS_H
+
+#include <stdint.h>
+
+/**
+ * Circularly convolve fixed vector with a phase dispersion impulse
+ * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
+ * @param fc_out vector with filter applied
+ * @param fc_in source vector
+ * @param filter phase filter coefficients
+ *
+ * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
+ *
+ * \note fc_in and fc_out should not overlap!
+ */
+void ff_celp_convolve_circ(
+ int16_t* fc_out,
+ const int16_t* fc_in,
+ const int16_t* filter,
+ int len);
+
+/**
+ * LP synthesis filter.
+ * @param out [out] pointer to output buffer
+ * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
+ * @param in input signal
+ * @param buffer_length amount of data to process
+ * @param filter_length filter length (10 for 10th order LP filter)
+ * @param stop_on_overflow 1 - return immediately if overflow occurs
+ * 0 - ignore overflows
+ * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
+ *
+ * @return 1 if overflow occurred, 0 - otherwise
+ *
+ * @note Output buffer must contain 10 samples of past
+ * speech data before pointer.
+ *
+ * Routine applies 1/A(z) filter to given speech data.
+ */
+int ff_celp_lp_synthesis_filter(
+ int16_t *out,
+ const int16_t* filter_coeffs,
+ const int16_t* in,
+ int buffer_length,
+ int filter_length,
+ int stop_on_overflow,
+ int rounder);
+
+#endif /* AVCODEC_CELP_FILTERS_H */
diff --git a/libavcodec/ra144.c b/libavcodec/ra144.c
index 83ad0b0..01cbc86 100644
--- a/libavcodec/ra144.c
+++ b/libavcodec/ra144.c
@@ -25,7 +25,7 @@
#include "avcodec.h"
#include "bitstream.h"
#include "ra144.h"
-#include "acelp_filters.h"
+#include "celp_filters.h"
#define NBLOCKS 4 ///< number of subblocks within a block
#define BLOCKSIZE 40 ///< subblock size in 16-bit words
@@ -201,8 +201,8 @@ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
10*sizeof(*ractx->curr_sblock));
- if (ff_acelp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
- block, BLOCKSIZE, 10, 1, 0xfff))
+ if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
+ block, BLOCKSIZE, 10, 1, 0xfff))
memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
}
--
Libav/FFmpeg packaging
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