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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.
This commit is contained in:
Stefano Sabatini 2011-07-04 11:35:39 +03:00 committed by Stefano Sabatini
parent 553c5e9f23
commit 37cc443c83
8 changed files with 632 additions and 1 deletions
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1
Changelog
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@ -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:

1
configure vendored
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@ -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"

36
doc/filters.texi
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@ -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

3
libavfilter/Makefile
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@ -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

417
libavfilter/af_aconvert.c Normal file
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@ -0,0 +1,417 @@
/*
* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@ucsd.edu>
* Copyright (c) 2011 Stefano Sabatini
* 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
* sample format and channel layout conversion audio filter
* based on code in libavcodec/resample.c by Fabrice Bellard and
* libavcodec/audioconvert.c by Michael Niedermayer
*/
#include "libavutil/audioconvert.h"
#include "libavcodec/audioconvert.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
enum AVSampleFormat out_sample_fmt, in_sample_fmt; ///< in/out sample formats
int64_t out_chlayout, in_chlayout; ///< in/out channel layout
int out_nb_channels, in_nb_channels; ///< number of in/output channels
enum AVFilterPacking out_packing_fmt, in_packing_fmt; ///< output packing format
int max_nb_samples; ///< maximum number of buffered samples
AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer
AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions
uint8_t *in_mix[8], *out_mix[8]; ///< input/output for rematrixing functions
uint8_t *packed_data[8]; ///< pointers for packing conversion
int out_strides[8], in_strides[8]; ///< input/output strides for av_audio_convert
uint8_t **in_conv, **out_conv; ///< input/output for av_audio_convert
AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format
void (*convert_chlayout)(); ///< function to do the requested rematrixing
} AConvertContext;
#define REMATRIX_FUNC_SIG(NAME) static void REMATRIX_FUNC_NAME(NAME) \
(FMT_TYPE *outp[], FMT_TYPE *inp[], int nb_samples, AConvertContext *aconvert)
#define FMT_TYPE uint8_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _u8
#include "af_aconvert_rematrix.c"
#define FMT_TYPE int16_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _s16
#include "af_aconvert_rematrix.c"
#define FMT_TYPE int32_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _s32
#include "af_aconvert_rematrix.c"
#define FLOATING
#define FMT_TYPE float
#define REMATRIX_FUNC_NAME(NAME) NAME ## _flt
#include "af_aconvert_rematrix.c"
#define FMT_TYPE double
#define REMATRIX_FUNC_NAME(NAME) NAME ## _dbl
#include "af_aconvert_rematrix.c"
#define FMT_TYPE uint8_t
#define REMATRIX_FUNC_NAME(NAME) NAME
REMATRIX_FUNC_SIG(stereo_remix_planar)
{
int size = av_get_bytes_per_sample(aconvert->in_sample_fmt) * nb_samples;
memcpy(outp[0], inp[0], size);
memcpy(outp[1], inp[aconvert->in_nb_channels == 1 ? 0 : 1], size);
}
#define REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC, PACKING) \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_U8, FUNC##_u8}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S16, FUNC##_s16}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S32, FUNC##_s32}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_FLT, FUNC##_flt}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_DBL, FUNC##_dbl},
#define REGISTER_FUNC(INCHLAYOUT, OUTCHLAYOUT, FUNC) \
REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_packed, AVFILTER_PACKED) \
REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_planar, AVFILTER_PLANAR)
static struct RematrixFunctionInfo {
int64_t in_chlayout, out_chlayout;
int planar, sfmt;
void (*func)();
} rematrix_funcs[] = {
REGISTER_FUNC (AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, stereo_to_surround_5p1)
REGISTER_FUNC (AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_5p1_to_stereo)
REGISTER_FUNC_PACKING(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, stereo_to_mono_packed, AVFILTER_PACKED)
REGISTER_FUNC_PACKING(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, mono_to_stereo_packed, AVFILTER_PACKED)
REGISTER_FUNC (0, AV_CH_LAYOUT_MONO, mono_downmix)
REGISTER_FUNC_PACKING(0, AV_CH_LAYOUT_STEREO, stereo_downmix_packed, AVFILTER_PACKED)
// This function works for all sample formats
{0, AV_CH_LAYOUT_STEREO, AVFILTER_PLANAR, -1, stereo_remix_planar}
};
static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
{
AConvertContext *aconvert = ctx->priv;
char *arg, *ptr = NULL;
int ret = 0;
char *args = av_strdup(args0);
aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE;
aconvert->out_chlayout = 0;
aconvert->out_packing_fmt = -1;
if ((arg = strtok_r(args, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0)
goto end;
}
if ((arg = strtok_r(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0)
goto end;
}
if ((arg = strtok_r(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_packing_format((int *)&aconvert->out_packing_fmt, arg, ctx)) < 0)
goto end;
}
end:
av_freep(&args);
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AConvertContext *aconvert = ctx->priv;
avfilter_unref_buffer(aconvert->mix_samplesref);
avfilter_unref_buffer(aconvert->out_samplesref);
if (aconvert->audioconvert_ctx)
av_audio_convert_free(aconvert->audioconvert_ctx);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AConvertContext *aconvert = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&inlink->out_formats);
if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_sample_fmt);
avfilter_formats_ref(formats, &outlink->in_formats);
} else
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&outlink->in_formats);
avfilter_formats_ref(avfilter_make_all_channel_layouts(),
&inlink->out_chlayouts);
if (aconvert->out_chlayout != 0) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_chlayout);
avfilter_formats_ref(formats, &outlink->in_chlayouts);
} else
avfilter_formats_ref(avfilter_make_all_channel_layouts(),
&outlink->in_chlayouts);
avfilter_formats_ref(avfilter_make_all_packing_formats(),
&inlink->out_packing);
if (aconvert->out_packing_fmt != -1) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_packing_fmt);
avfilter_formats_ref(formats, &outlink->in_packing);
} else
avfilter_formats_ref(avfilter_make_all_packing_formats(),
&outlink->in_packing);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterLink *inlink = outlink->src->inputs[0];
AConvertContext *aconvert = outlink->src->priv;
char buf1[64], buf2[64];
aconvert->in_sample_fmt = inlink->format;
aconvert->in_packing_fmt = inlink->planar;
if (aconvert->out_packing_fmt == -1)
aconvert->out_packing_fmt = outlink->planar;
aconvert->in_chlayout = inlink->channel_layout;
aconvert->in_nb_channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
/* if not specified in args, use the format and layout of the output */
if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE)
aconvert->out_sample_fmt = outlink->format;
if (aconvert->out_chlayout == 0)
aconvert->out_chlayout = outlink->channel_layout;
aconvert->out_nb_channels =
av_get_channel_layout_nb_channels(outlink->channel_layout);
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(outlink->src, AV_LOG_INFO,
"fmt:%s cl:%s planar:%i -> fmt:%s cl:%s planar:%i\n",
av_get_sample_fmt_name(inlink ->format), buf1, inlink ->planar,
av_get_sample_fmt_name(outlink->format), buf2, outlink->planar);
/* compute which channel layout conversion to use */
if (inlink->channel_layout != outlink->channel_layout) {
int i;
for (i = 0; i < sizeof(rematrix_funcs); i++) {
const struct RematrixFunctionInfo *f = &rematrix_funcs[i];
if ((f->in_chlayout == 0 || f->in_chlayout == inlink ->channel_layout) &&
(f->out_chlayout == 0 || f->out_chlayout == outlink->channel_layout) &&
(f->planar == -1 || f->planar == inlink->planar) &&
(f->sfmt == -1 || f->sfmt == inlink->format)
) {
aconvert->convert_chlayout = f->func;
break;
}
}
if (!aconvert->convert_chlayout) {
av_log(outlink->src, AV_LOG_ERROR,
"Unsupported channel layout conversion '%s -> %s' requested!\n",
buf1, buf2);
return AVERROR(EINVAL);
}
}
return 0;
}
static int init_buffers(AVFilterLink *inlink, int nb_samples)
{
AConvertContext *aconvert = inlink->dst->priv;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int i, packed_stride = 0;
const unsigned
packing_conv = inlink->planar != outlink->planar &&
aconvert->out_nb_channels != 1,
format_conv = inlink->format != outlink->format;
int nb_channels = aconvert->out_nb_channels;
uninit(inlink->dst);
aconvert->max_nb_samples = nb_samples;
if (aconvert->convert_chlayout) {
/* allocate buffer for storing intermediary mixing samplesref */
uint8_t *data[8];
int linesize[8];
int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
if (av_samples_alloc(data, linesize, nb_channels, nb_samples,
inlink->format, inlink->planar, 16) < 0)
goto fail_no_mem;
aconvert->mix_samplesref =
avfilter_get_audio_buffer_ref_from_arrays(data, linesize, AV_PERM_WRITE,
nb_samples, inlink->format,
outlink->channel_layout,
inlink->planar);
if (!aconvert->mix_samplesref)
goto fail_no_mem;
}
// if there's a format/packing conversion we need an audio_convert context
if (format_conv || packing_conv) {
aconvert->out_samplesref =
avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!aconvert->out_samplesref)
goto fail_no_mem;
aconvert->in_strides [0] = av_get_bytes_per_sample(inlink ->format);
aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format);
aconvert->out_conv = aconvert->out_samplesref->data;
if (aconvert->mix_samplesref)
aconvert->in_conv = aconvert->mix_samplesref->data;
if (packing_conv) {
// packed -> planar
if (outlink->planar == AVFILTER_PLANAR) {
if (aconvert->mix_samplesref)
aconvert->packed_data[0] = aconvert->mix_samplesref->data[0];
aconvert->in_conv = aconvert->packed_data;
packed_stride = aconvert->in_strides[0];
aconvert->in_strides[0] *= nb_channels;
// planar -> packed
} else {
aconvert->packed_data[0] = aconvert->out_samplesref->data[0];
aconvert->out_conv = aconvert->packed_data;
packed_stride = aconvert->out_strides[0];
aconvert->out_strides[0] *= nb_channels;
}
} else if (outlink->planar == AVFILTER_PACKED) {
/* If there's no packing conversion, and the stream is packed
* then we treat the entire stream as one big channel
*/
nb_channels = 1;
}
for (i = 1; i < nb_channels; i++) {
aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
aconvert->in_strides[i] = aconvert->in_strides[0];
aconvert->out_strides[i] = aconvert->out_strides[0];
}
aconvert->audioconvert_ctx =
av_audio_convert_alloc(outlink->format, nb_channels,
inlink->format, nb_channels, NULL, 0);
if (!aconvert->audioconvert_ctx)
goto fail_no_mem;
}
return 0;
fail_no_mem:
av_log(inlink->dst, AV_LOG_ERROR, "Could not allocate memory.\n");
return AVERROR(ENOMEM);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
AVFilterBufferRef *curbuf = insamplesref;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int chan_mult;
/* in/reinint the internal buffers if this is the first buffer
* provided or it is needed to use a bigger one */
if (!aconvert->max_nb_samples ||
(curbuf->audio->nb_samples > aconvert->max_nb_samples))
if (init_buffers(inlink, curbuf->audio->nb_samples) < 0) {
av_log(inlink->dst, AV_LOG_ERROR, "Could not initialize buffers.\n");
return;
}
/* if channel mixing is required */
if (aconvert->mix_samplesref) {
memcpy(aconvert->in_mix, curbuf->data, sizeof(aconvert->in_mix));
memcpy(aconvert->out_mix, aconvert->mix_samplesref->data, sizeof(aconvert->out_mix));
aconvert->convert_chlayout(aconvert->out_mix,
aconvert->in_mix,
curbuf->audio->nb_samples,
aconvert);
curbuf = aconvert->mix_samplesref;
}
if (aconvert->audioconvert_ctx) {
if (!aconvert->mix_samplesref) {
if (aconvert->in_conv == aconvert->packed_data) {
int i, packed_stride = av_get_bytes_per_sample(inlink->format);
aconvert->packed_data[0] = curbuf->data[0];
for (i = 1; i < aconvert->out_nb_channels; i++)
aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
} else {
aconvert->in_conv = curbuf->data;
}
}
chan_mult = inlink->planar == outlink->planar && inlink->planar == 0 ?
aconvert->out_nb_channels : 1;
av_audio_convert(aconvert->audioconvert_ctx,
(void * const *) aconvert->out_conv,
aconvert->out_strides,
(const void * const *) aconvert->in_conv,
aconvert->in_strides,
curbuf->audio->nb_samples * chan_mult);
curbuf = aconvert->out_samplesref;
}
avfilter_copy_buffer_ref_props(curbuf, insamplesref);
curbuf->audio->channel_layout = outlink->channel_layout;
curbuf->audio->planar = outlink->planar;
avfilter_filter_samples(inlink->dst->outputs[0],
avfilter_ref_buffer(curbuf, ~0));
avfilter_unref_buffer(insamplesref);
}
AVFilter avfilter_af_aconvert = {
.name = "aconvert",
.description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout:packed_fmt."),
.priv_size = sizeof(AConvertContext),
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output, },
{ .name = NULL}},
};

172
libavfilter/af_aconvert_rematrix.c Normal file
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@ -0,0 +1,172 @@
/*
* 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

1
libavfilter/allfilters.c
View File

@ -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);

2
libavfilter/avfilter.h
View File

@ -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, \