lavfi: add audio convert filter

Add aconvert filter to perform sample format, channel layout, and
packing format conversion.

The aconvert code depends on audio conversion code in libavcodec, so
this requires a dependency on libavcodec.

Based on previous work by S.N. Hemanth Meenakshisundaram and Mina Nagy
Zaki, performed for the GSoC 2010 and 2011.

Changelog

@@ -55,6 +55,7 @@ easier to use. The changes are:
 - H.264 Decoding on Android via Stagefright
 - Prores decoder
 - BIN/XBIN/ADF/IDF text file decoder
+- aconvert audio filter added
 
 
 version 0.8:

configure

@@ -1575,6 +1575,7 @@ udp_protocol_deps="network"
 
 # filters
 abuffer_filter_deps="strtok_r"
+aconvert_filter_deps="strtok_r"
 aformat_filter_deps="strtok_r"
 amovie_filter_deps="avcodec avformat"
 blackframe_filter_deps="gpl"

doc/filters.texi

@@ -99,6 +99,42 @@ build.
 
 Below is a description of the currently available audio filters.
 
+@section aconvert
+
+Convert the input audio format to the specified formats.
+
+The filter accepts a string of the form:
+"@var{sample_format}:@var{channel_layout}:@var{packing_format}".
+
+@var{sample_format} specifies the sample format, and can be a string or
+the corresponding numeric value defined in @file{libavutil/samplefmt.h}.
+
+@var{channel_layout} specifies the channel layout, and can be a string
+or the corresponding numer value defined in @file{libavutil/chlayout.h}.
+
+@var{packing_format} specifies the type of packing in output, can be one
+of "planar" or "packed", or the corresponding numeric values "0" or "1".
+
+The special parameter "auto", signifies that the filter will
+automatically select the output format depending on the output filter.
+
+Some examples follow.
+
+@itemize
+@item
+Convert input to unsigned 8-bit, stereo, packed:
+@example
+aconvert=u8:stereo:packed
+@end example
+
+@item
+Convert input to unsigned 8-bit, automatically select out channel layout
+and packing format:
+@example
+aconvert=u8:auto:auto
+@end example
+@end itemize
+
 @section aformat
 
 Convert the input audio to one of the specified formats. The framework will

libavfilter/Makefile

@@ -2,6 +2,8 @@ include $(SUBDIR)../config.mak
 
 NAME = avfilter
 FFLIBS = avutil
+
+FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
 FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
 FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += avcodec
 FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
@@ -20,6 +22,7 @@ OBJS = allfilters.o                                                     \
 
 OBJS-$(CONFIG_AVCODEC)                       += avcodec.o
 
+OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o

libavfilter/af_aconvert_rematrix.c

+/*
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * audio rematrixing functions, based on functions from libavcodec/resample.c
+ */
+
+#if defined(FLOATING)
+# define DIV2 /2
+#else
+# define DIV2 >>1
+#endif
+
+REMATRIX_FUNC_SIG(stereo_to_mono_packed)
+{
+    while (nb_samples >= 4) {
+        outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
+        outp[0][1] = (inp[0][2] + inp[0][3]) DIV2;
+        outp[0][2] = (inp[0][4] + inp[0][5]) DIV2;
+        outp[0][3] = (inp[0][6] + inp[0][7]) DIV2;
+        outp[0] += 4;
+        inp[0]  += 8;
+        nb_samples -= 4;
+    }
+    while (nb_samples--) {
+        outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
+        outp[0]++;
+        inp[0] += 2;
+    }
+}
+
+REMATRIX_FUNC_SIG(stereo_downmix_packed)
+{
+    while (nb_samples--) {
+        *outp[0]++ = inp[0][0];
+        *outp[0]++ = inp[0][1];
+        inp[0] += aconvert->in_nb_channels;
+    }
+}
+
+REMATRIX_FUNC_SIG(mono_to_stereo_packed)
+{
+    while (nb_samples >= 4) {
+        outp[0][0] = outp[0][1] = inp[0][0];
+        outp[0][2] = outp[0][3] = inp[0][1];
+        outp[0][4] = outp[0][5] = inp[0][2];
+        outp[0][6] = outp[0][7] = inp[0][3];
+        outp[0] += 8;
+        inp[0]  += 4;
+        nb_samples -= 4;
+    }
+    while (nb_samples--) {
+        outp[0][0] = outp[0][1] = inp[0][0];
+        outp[0] += 2;
+        inp[0]  += 1;
+    }
+}
+
+/**
+ * This is for when we have more than 2 input channels, need to downmix to mono
+ * and do not have a conversion formula available.  We just use first two input
+ * channels - left and right. This is a placeholder until more conversion
+ * functions are written.
+ */
+REMATRIX_FUNC_SIG(mono_downmix_packed)
+{
+    while (nb_samples--) {
+        outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
+        inp[0] += aconvert->in_nb_channels;
+        outp[0]++;
+    }
+}
+
+REMATRIX_FUNC_SIG(mono_downmix_planar)
+{
+    FMT_TYPE *out = outp[0];
+
+    while (nb_samples >= 4) {
+        out[0] = (inp[0][0] + inp[1][0]) DIV2;
+        out[1] = (inp[0][1] + inp[1][1]) DIV2;
+        out[2] = (inp[0][2] + inp[1][2]) DIV2;
+        out[3] = (inp[0][3] + inp[1][3]) DIV2;
+        out    += 4;
+        inp[0] += 4;
+        inp[1] += 4;
+        nb_samples -= 4;
+    }
+    while (nb_samples--) {
+        out[0] = (inp[0][0] + inp[1][0]) DIV2;
+        out++;
+        inp[0]++;
+        inp[1]++;
+    }
+}
+
+/* Stereo to 5.1 output */
+REMATRIX_FUNC_SIG(stereo_to_surround_5p1_packed)
+{
+    while (nb_samples--) {
+      outp[0][0] = inp[0][0];  /* left */
+      outp[0][1] = inp[0][1];  /* right */
+      outp[0][2] = (inp[0][0] + inp[0][1]) DIV2; /* center */
+      outp[0][3] = 0;          /* low freq */
+      outp[0][4] = 0;          /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left  */
+      outp[0][5] = 0;          /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
+      inp[0]  += 2;
+      outp[0] += 6;
+    }
+}
+
+REMATRIX_FUNC_SIG(stereo_to_surround_5p1_planar)
+{
+    while (nb_samples--) {
+      *outp[0]++ = *inp[0];    /* left */
+      *outp[1]++ = *inp[1];    /* right */
+      *outp[2]++ = (*inp[0] + *inp[1]) DIV2; /* center */
+      *outp[3]++ = 0;          /* low freq */
+      *outp[4]++ = 0;          /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left  */
+      *outp[5]++ = 0;          /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
+      inp[0]++; inp[1]++;
+    }
+}
+
+
+/*
+5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
+- Left = front_left + rear_gain * rear_left + center_gain * center
+- Right = front_right + rear_gain * rear_right + center_gain * center
+Where rear_gain is usually around 0.5-1.0 and
+      center_gain is almost always 0.7 (-3 dB)
+*/
+REMATRIX_FUNC_SIG(surround_5p1_to_stereo_packed)
+{
+    while (nb_samples--) {
+        *outp[0]++ = inp[0][0] + (0.5 * inp[0][4]) + (0.7 * inp[0][2]); //FIXME CLIPPING!
+        *outp[0]++ = inp[0][1] + (0.5 * inp[0][5]) + (0.7 * inp[0][2]); //FIXME CLIPPING!
+
+        inp[0] += 6;
+    }
+}
+
+REMATRIX_FUNC_SIG(surround_5p1_to_stereo_planar)
+{
+    while (nb_samples--) {
+        *outp[0]++ = *inp[0] + (0.5 * *inp[4]) + (0.7 * *inp[2]); //FIXME CLIPPING!
+        *outp[1]++ = *inp[1] + (0.5 * *inp[5]) + (0.7 * *inp[2]); //FIXME CLIPPING!
+
+        inp[0]++; inp[1]++; inp[2]++; inp[3]++; inp[4]++; inp[5]++;
+    }
+}
+
+#undef DIV2
+#undef REMATRIX_FUNC_NAME
+#undef FMT_TYPE

libavfilter/allfilters.c

@@ -34,6 +34,7 @@ void avfilter_register_all(void)
         return;
     initialized = 1;
 
+    REGISTER_FILTER (ACONVERT,    aconvert,    af);
     REGISTER_FILTER (AFORMAT,     aformat,     af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);

libavfilter/avfilter.h

@@ -29,7 +29,7 @@
 #include "libavutil/rational.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 42
+#define LIBAVFILTER_VERSION_MINOR 43
 #define LIBAVFILTER_VERSION_MICRO  0
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \