[FFmpeg-cvslog] lavfi/aconvert: use libswresample.
Clément Bœsch
git at videolan.org
Thu Feb 2 14:41:13 CET 2012
ffmpeg | branch: master | Clément Bœsch <ubitux at gmail.com> | Thu Jan 26 08:49:15 2012 +0100| [c79eddaff16492fe7eb5751d2101aebedc9d16cf] | committer: Clément Bœsch
lavfi/aconvert: use libswresample.
This commit also drops the planar parameter; you now need to use the 'p'
suffix in order to request a planar sample format.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c79eddaff16492fe7eb5751d2101aebedc9d16cf
---
configure | 1 +
doc/filters.texi | 19 +--
libavfilter/Makefile | 2 +-
libavfilter/af_aconvert.c | 307 ++++--------------------------------
libavfilter/af_aconvert_rematrix.c | 172 --------------------
libavfilter/version.h | 2 +-
6 files changed, 44 insertions(+), 459 deletions(-)
diff --git a/configure b/configure
index b4e433c..aa45fd7 100755
--- a/configure
+++ b/configure
@@ -1648,6 +1648,7 @@ tls_protocol_select="tcp_protocol"
udp_protocol_deps="network"
# filters
+aconvert_filter_deps="swresample"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
diff --git a/doc/filters.texi b/doc/filters.texi
index 1f3522e..c64f899 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -104,17 +104,15 @@ Below is a description of the currently available audio filters.
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}:@var{channel_layout}".
- at var{sample_format} specifies the sample format, and can be a string or
-the corresponding numeric value defined in @file{libavutil/samplefmt.h}.
+ at var{sample_format} specifies the sample format, and can be a string or the
+corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p'
+suffix for a planar sample format.
@var{channel_layout} specifies the channel layout, and can be a string
or the corresponding number value defined in @file{libavutil/audioconvert.h}.
- at 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.
@@ -122,16 +120,15 @@ Some examples follow.
@itemize
@item
-Convert input to unsigned 8-bit, stereo, packed:
+Convert input to float, planar, stereo:
@example
-aconvert=u8:stereo:packed
+aconvert=fltp:stereo
@end example
@item
-Convert input to unsigned 8-bit, automatically select out channel layout
-and packing format:
+Convert input to unsigned 8-bit, automatically select out channel layout:
@example
-aconvert=u8:auto:auto
+aconvert=u8:auto
@end example
@end itemize
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index ff0ba75..9fbb59b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -3,7 +3,7 @@ include $(SUBDIR)../config.mak
NAME = avfilter
FFLIBS = avutil
-FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
+FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
diff --git a/libavfilter/af_aconvert.c b/libavfilter/af_aconvert.c
index e3c7f8c..8c1b5dc 100644
--- a/libavfilter/af_aconvert.c
+++ b/libavfilter/af_aconvert.c
@@ -23,98 +23,19 @@
/**
* @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 "libavutil/avstring.h"
-#include "libavcodec/audioconvert.h"
+#include "libswresample/swresample.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
+ enum AVSampleFormat out_sample_fmt;
+ int64_t out_chlayout;
+ struct SwrContext *swr;
} 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 const 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;
@@ -124,7 +45,6 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE;
aconvert->out_chlayout = 0;
- aconvert->out_packing_fmt = -1;
if ((arg = av_strtok(args, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0)
@@ -134,10 +54,6 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0)
goto end;
}
- if ((arg = av_strtok(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);
@@ -147,10 +63,7 @@ end:
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);
+ swr_free(&aconvert->swr);
}
static int query_formats(AVFilterContext *ctx)
@@ -159,6 +72,7 @@ static int query_formats(AVFilterContext *ctx)
AConvertContext *aconvert = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
+ int out_packing = av_sample_fmt_is_planar(aconvert->out_sample_fmt);
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&inlink->out_formats);
@@ -182,219 +96,64 @@ static int query_formats(AVFilterContext *ctx)
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);
+ formats = NULL;
+ avfilter_add_format(&formats, out_packing);
+ avfilter_formats_ref(formats, &outlink->in_packing);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
- AVFilterLink *inlink = outlink->src->inputs[0];
- AConvertContext *aconvert = outlink->src->priv;
+ int ret;
+ AVFilterContext *ctx = outlink->src;
+ AVFilterLink *inlink = ctx->inputs[0];
+ AConvertContext *aconvert = ctx->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);
+
+ aconvert->swr = swr_alloc_set_opts(aconvert->swr,
+ aconvert->out_chlayout, aconvert->out_sample_fmt, inlink->sample_rate,
+ inlink->channel_layout, inlink->format, inlink->sample_rate,
+ 0, ctx);
+ if (!aconvert->swr)
+ return AVERROR(ENOMEM);
+ ret = swr_init(aconvert->swr);
+ if (ret < 0)
+ return ret;
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,
+ av_log(ctx, 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, 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;
- }
+ const int n = insamplesref->audio->nb_samples;
+ AVFilterLink *const outlink = inlink->dst->outputs[0];
+ AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n);
- /* 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;
- }
+ swr_convert(aconvert->swr, outsamplesref->data, n,
+ (void *)insamplesref->data, n);
- avfilter_copy_buffer_ref_props(curbuf, insamplesref);
- curbuf->audio->channel_layout = outlink->channel_layout;
- curbuf->audio->planar = outlink->planar;
+ avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
+ outsamplesref->audio->channel_layout = outlink->channel_layout;
+ outsamplesref->audio->planar = outlink->planar;
- avfilter_filter_samples(inlink->dst->outputs[0],
- avfilter_ref_buffer(curbuf, ~0));
+ avfilter_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
}
diff --git a/libavfilter/af_aconvert_rematrix.c b/libavfilter/af_aconvert_rematrix.c
deleted file mode 100644
index d75ca5a..0000000
--- a/libavfilter/af_aconvert_rematrix.c
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- * 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
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 393fb91..11e038d 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 2
-#define LIBAVFILTER_VERSION_MINOR 60
+#define LIBAVFILTER_VERSION_MINOR 61
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
More information about the ffmpeg-cvslog
mailing list