[FFmpeg-devel] [PATCH 72/87] libavresample: Remove deprecated library
James Almer
jamrial at gmail.com
Mon Apr 19 17:10:09 EEST 2021
From: Andreas Rheinhardt <andreas.rheinhardt at outlook.com>
Deprecated in c29038f3041a4080342b2e333c1967d136749c0f.
The resample filter based upon this library has been removed as well.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt at outlook.com>
---
Makefile | 1 -
configure | 16 +-
ffbuild/common.mak | 2 +-
fftools/cmdutils.c | 12 -
fftools/ffmpeg_filter.c | 2 -
libavfilter/Makefile | 1 -
libavfilter/af_resample.c | 369 ------
libavfilter/allfilters.c | 1 -
libavresample/Makefile | 19 -
libavresample/aarch64/Makefile | 7 -
libavresample/aarch64/asm-offsets.h | 28 -
libavresample/aarch64/audio_convert_init.c | 49 -
libavresample/aarch64/audio_convert_neon.S | 363 ------
libavresample/aarch64/neontest.c | 31 -
libavresample/aarch64/resample_init.c | 71 --
libavresample/aarch64/resample_neon.S | 233 ----
libavresample/arm/Makefile | 7 -
libavresample/arm/asm-offsets.h | 29 -
libavresample/arm/audio_convert_init.c | 49 -
libavresample/arm/audio_convert_neon.S | 363 ------
libavresample/arm/neontest.c | 31 -
libavresample/arm/resample_init.c | 74 --
libavresample/arm/resample_neon.S | 358 ------
libavresample/audio_convert.c | 416 -------
libavresample/audio_convert.h | 103 --
libavresample/audio_data.c | 381 ------
libavresample/audio_data.h | 178 ---
libavresample/audio_mix.c | 742 ------------
libavresample/audio_mix.h | 94 --
libavresample/audio_mix_matrix.c | 294 -----
libavresample/avresample.h | 595 ---------
libavresample/avresampleres.rc | 55 -
libavresample/dither.c | 440 -------
libavresample/dither.h | 93 --
libavresample/internal.h | 116 --
libavresample/libavresample.v | 6 -
libavresample/options.c | 113 --
libavresample/resample.c | 446 -------
libavresample/resample.h | 96 --
libavresample/resample_template.c | 118 --
libavresample/tests/.gitignore | 1 -
libavresample/tests/avresample.c | 342 ------
libavresample/utils.c | 793 ------------
libavresample/version.h | 50 -
libavresample/x86/Makefile | 9 -
libavresample/x86/audio_convert.asm | 1261 --------------------
libavresample/x86/audio_convert_init.c | 265 ----
libavresample/x86/audio_mix.asm | 511 --------
libavresample/x86/audio_mix_init.c | 215 ----
libavresample/x86/dither.asm | 117 --
libavresample/x86/dither_init.c | 60 -
libavresample/x86/util.asm | 41 -
libavresample/x86/w64xmmtest.c | 31 -
tests/Makefile | 1 -
tests/fate.sh | 1 -
tests/fate/libavresample.mak | 68 --
tools/gen-rc | 1 -
tools/target_dec_fuzzer.c | 2 +-
58 files changed, 5 insertions(+), 10166 deletions(-)
delete mode 100644 libavfilter/af_resample.c
delete mode 100644 libavresample/Makefile
delete mode 100644 libavresample/aarch64/Makefile
delete mode 100644 libavresample/aarch64/asm-offsets.h
delete mode 100644 libavresample/aarch64/audio_convert_init.c
delete mode 100644 libavresample/aarch64/audio_convert_neon.S
delete mode 100644 libavresample/aarch64/neontest.c
delete mode 100644 libavresample/aarch64/resample_init.c
delete mode 100644 libavresample/aarch64/resample_neon.S
delete mode 100644 libavresample/arm/Makefile
delete mode 100644 libavresample/arm/asm-offsets.h
delete mode 100644 libavresample/arm/audio_convert_init.c
delete mode 100644 libavresample/arm/audio_convert_neon.S
delete mode 100644 libavresample/arm/neontest.c
delete mode 100644 libavresample/arm/resample_init.c
delete mode 100644 libavresample/arm/resample_neon.S
delete mode 100644 libavresample/audio_convert.c
delete mode 100644 libavresample/audio_convert.h
delete mode 100644 libavresample/audio_data.c
delete mode 100644 libavresample/audio_data.h
delete mode 100644 libavresample/audio_mix.c
delete mode 100644 libavresample/audio_mix.h
delete mode 100644 libavresample/audio_mix_matrix.c
delete mode 100644 libavresample/avresample.h
delete mode 100644 libavresample/avresampleres.rc
delete mode 100644 libavresample/dither.c
delete mode 100644 libavresample/dither.h
delete mode 100644 libavresample/internal.h
delete mode 100644 libavresample/libavresample.v
delete mode 100644 libavresample/options.c
delete mode 100644 libavresample/resample.c
delete mode 100644 libavresample/resample.h
delete mode 100644 libavresample/resample_template.c
delete mode 100644 libavresample/tests/.gitignore
delete mode 100644 libavresample/tests/avresample.c
delete mode 100644 libavresample/utils.c
delete mode 100644 libavresample/version.h
delete mode 100644 libavresample/x86/Makefile
delete mode 100644 libavresample/x86/audio_convert.asm
delete mode 100644 libavresample/x86/audio_convert_init.c
delete mode 100644 libavresample/x86/audio_mix.asm
delete mode 100644 libavresample/x86/audio_mix_init.c
delete mode 100644 libavresample/x86/dither.asm
delete mode 100644 libavresample/x86/dither_init.c
delete mode 100644 libavresample/x86/util.asm
delete mode 100644 libavresample/x86/w64xmmtest.c
delete mode 100644 tests/fate/libavresample.mak
diff --git a/Makefile b/Makefile
index 7e9d8b08c3..1e3da6271b 100644
--- a/Makefile
+++ b/Makefile
@@ -23,7 +23,6 @@ FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
diff --git a/configure b/configure
index 30b598dcc6..d14eb552ea 100755
--- a/configure
+++ b/configure
@@ -132,7 +132,6 @@ Component options:
--disable-swscale disable libswscale build
--disable-postproc disable libpostproc build
--disable-avfilter disable libavfilter build
- --enable-avresample enable libavresample build (deprecated) [no]
--disable-pthreads disable pthreads [autodetect]
--disable-w32threads disable Win32 threads [autodetect]
--disable-os2threads disable OS/2 threads [autodetect]
@@ -1901,7 +1900,6 @@ LIBRARY_LIST="
avformat
avcodec
swresample
- avresample
avutil
"
@@ -3612,7 +3610,6 @@ program_opencl_filter_deps="opencl"
pullup_filter_deps="gpl"
removelogo_filter_deps="avcodec avformat swscale"
repeatfields_filter_deps="gpl"
-resample_filter_deps="avresample"
roberts_opencl_filter_deps="opencl"
rubberband_filter_deps="librubberband"
sab_filter_deps="gpl swscale"
@@ -3714,8 +3711,6 @@ avfilter_deps="avutil"
avfilter_suggest="libm"
avformat_deps="avcodec avutil"
avformat_suggest="libm network zlib"
-avresample_deps="avutil"
-avresample_suggest="libm"
avutil_suggest="clock_gettime ffnvcodec libm libdrm libmfx opencl user32 vaapi vulkan videotoolbox corefoundation corevideo coremedia bcrypt"
postproc_deps="avutil gpl"
postproc_suggest="libm"
@@ -3798,7 +3793,7 @@ intrinsics="none"
enable $PROGRAM_LIST
enable $DOCUMENT_LIST
enable $EXAMPLE_LIST
-enable $(filter_out avresample $LIBRARY_LIST)
+enable $LIBRARY_LIST
enable stripping
enable asm
@@ -6848,7 +6843,7 @@ EOF
# add some linker flags
check_ldflags -Wl,--warn-common
-check_ldflags -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample
+check_ldflags -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil
enabled rpath && add_ldexeflags -Wl,-rpath,$libdir && add_ldsoflags -Wl,-rpath,$libdir
test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic
@@ -6863,8 +6858,7 @@ enabled neon_clobber_test &&
-Wl,--wrap,avcodec_receive_packet \
-Wl,--wrap,avcodec_send_frame \
-Wl,--wrap,avcodec_receive_frame \
- -Wl,--wrap,swr_convert \
- -Wl,--wrap,avresample_convert ||
+ -Wl,--wrap,swr_convert ||
disable neon_clobber_test
enabled xmm_clobber_test &&
@@ -6876,7 +6870,6 @@ enabled xmm_clobber_test &&
-Wl,--wrap,avcodec_send_frame \
-Wl,--wrap,avcodec_receive_frame \
-Wl,--wrap,swr_convert \
- -Wl,--wrap,avresample_convert \
-Wl,--wrap,sws_scale ||
disable xmm_clobber_test
@@ -7101,7 +7094,6 @@ check_deps $CONFIG_LIST \
$ALL_COMPONENTS \
enabled threads && ! enabled pthreads && ! enabled atomics_native && die "non pthread threading without atomics not supported, try adding --enable-pthreads or --cpu=i486 or higher if you are on x86"
-enabled avresample && warn "Building with deprecated library libavresample"
case $target_os in
haiku)
@@ -7217,7 +7209,6 @@ enabled movie_filter && prepend avfilter_deps "avformat avcodec"
enabled pan_filter && prepend avfilter_deps "swresample"
enabled pp_filter && prepend avfilter_deps "postproc"
enabled removelogo_filter && prepend avfilter_deps "avformat avcodec swscale"
-enabled resample_filter && prepend avfilter_deps "avresample"
enabled sab_filter && prepend avfilter_deps "swscale"
enabled scale_filter && prepend avfilter_deps "swscale"
enabled scale2ref_filter && prepend avfilter_deps "swscale"
@@ -7710,7 +7701,6 @@ extralibs_avcodec="$avcodec_extralibs"
extralibs_avformat="$avformat_extralibs"
extralibs_avdevice="$avdevice_extralibs"
extralibs_avfilter="$avfilter_extralibs"
-extralibs_avresample="$avresample_extralibs"
extralibs_postproc="$postproc_extralibs"
extralibs_swscale="$swscale_extralibs"
extralibs_swresample="$swresample_extralibs"
diff --git a/ffbuild/common.mak b/ffbuild/common.mak
index e070b6b5e2..32f5b997b5 100644
--- a/ffbuild/common.mak
+++ b/ffbuild/common.mak
@@ -26,7 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_LINK)/
diff --git a/fftools/cmdutils.c b/fftools/cmdutils.c
index fe424b6a4c..1db5e8cdd9 100644
--- a/fftools/cmdutils.c
+++ b/fftools/cmdutils.c
@@ -34,7 +34,6 @@
#include "libavformat/avformat.h"
#include "libavfilter/avfilter.h"
#include "libavdevice/avdevice.h"
-#include "libavresample/avresample.h"
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
#include "libpostproc/postprocess.h"
@@ -545,9 +544,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
char opt_stripped[128];
const char *p;
const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class();
-#if CONFIG_AVRESAMPLE
- const AVClass *rc = avresample_get_class();
-#endif
#if CONFIG_SWSCALE
const AVClass *sc = sws_get_class();
#endif
@@ -617,13 +613,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
consumed = 1;
}
#endif
-#if CONFIG_AVRESAMPLE
- if ((o=opt_find(&rc, opt, NULL, 0,
- AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
- av_dict_set(&resample_opts, opt, arg, FLAGS);
- consumed = 1;
- }
-#endif
if (consumed)
return 0;
@@ -1134,7 +1123,6 @@ static void print_all_libs_info(int flags, int level)
PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
- PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
PRINT_LIB_INFO(swscale, SWSCALE, flags, level);
PRINT_LIB_INFO(swresample, SWRESAMPLE, flags, level);
PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
diff --git a/fftools/ffmpeg_filter.c b/fftools/ffmpeg_filter.c
index e7c05eb3f9..958d74f008 100644
--- a/fftools/ffmpeg_filter.c
+++ b/fftools/ffmpeg_filter.c
@@ -26,8 +26,6 @@
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
-#include "libavresample/avresample.h"
-
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 42efa14a67..5a287364b0 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -135,7 +135,6 @@ OBJS-$(CONFIG_LV2_FILTER) += af_lv2.o
OBJS-$(CONFIG_MCOMPAND_FILTER) += af_mcompand.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o
-OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_RUBBERBAND_FILTER) += af_rubberband.o
OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
deleted file mode 100644
index caa97d8ab0..0000000000
--- a/libavfilter/af_resample.c
+++ /dev/null
@@ -1,369 +0,0 @@
-/*
- * 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
- */
-
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/common.h"
-#include "libavutil/dict.h"
-#include "libavutil/mathematics.h"
-#include "libavutil/opt.h"
-
-#include "libavresample/avresample.h"
-
-#include "audio.h"
-#include "avfilter.h"
-#include "formats.h"
-#include "internal.h"
-
-typedef struct ResampleContext {
- const AVClass *class;
- AVAudioResampleContext *avr;
- AVDictionary *options;
-
- int resampling;
- int64_t next_pts;
- int64_t next_in_pts;
-
- /* set by filter_frame() to signal an output frame to request_frame() */
- int got_output;
-} ResampleContext;
-
-static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
-{
- ResampleContext *s = ctx->priv;
- const AVClass *avr_class = avresample_get_class();
- AVDictionaryEntry *e = NULL;
-
- while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
- if (av_opt_find(&avr_class, e->key, NULL, 0,
- AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
- av_dict_set(&s->options, e->key, e->value, 0);
- }
-
- e = NULL;
- while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
- av_dict_set(opts, e->key, NULL, 0);
-
- /* do not allow the user to override basic format options */
- av_dict_set(&s->options, "in_channel_layout", NULL, 0);
- av_dict_set(&s->options, "out_channel_layout", NULL, 0);
- av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
- av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
- av_dict_set(&s->options, "in_sample_rate", NULL, 0);
- av_dict_set(&s->options, "out_sample_rate", NULL, 0);
-
- return 0;
-}
-
-static av_cold void uninit(AVFilterContext *ctx)
-{
- ResampleContext *s = ctx->priv;
-
- if (s->avr) {
- avresample_close(s->avr);
- avresample_free(&s->avr);
- }
- av_dict_free(&s->options);
-}
-
-static int query_formats(AVFilterContext *ctx)
-{
- AVFilterLink *inlink = ctx->inputs[0];
- AVFilterLink *outlink = ctx->outputs[0];
- AVFilterFormats *in_formats, *out_formats, *in_samplerates, *out_samplerates;
- AVFilterChannelLayouts *in_layouts, *out_layouts;
- int ret;
-
- if (!(in_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) ||
- !(out_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) ||
- !(in_samplerates = ff_all_samplerates ( )) ||
- !(out_samplerates = ff_all_samplerates ( )) ||
- !(in_layouts = ff_all_channel_layouts ( )) ||
- !(out_layouts = ff_all_channel_layouts ( )))
- return AVERROR(ENOMEM);
-
- if ((ret = ff_formats_ref (in_formats, &inlink->outcfg.formats )) < 0 ||
- (ret = ff_formats_ref (out_formats, &outlink->incfg.formats )) < 0 ||
- (ret = ff_formats_ref (in_samplerates, &inlink->outcfg.samplerates )) < 0 ||
- (ret = ff_formats_ref (out_samplerates, &outlink->incfg.samplerates )) < 0 ||
- (ret = ff_channel_layouts_ref (in_layouts, &inlink->outcfg.channel_layouts)) < 0 ||
- (ret = ff_channel_layouts_ref (out_layouts, &outlink->incfg.channel_layouts)) < 0)
- return ret;
-
- return 0;
-}
-
-static int config_output(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- AVFilterLink *inlink = ctx->inputs[0];
- ResampleContext *s = ctx->priv;
- char buf1[64], buf2[64];
- int ret;
-
- int64_t resampling_forced;
-
- if (s->avr) {
- avresample_close(s->avr);
- avresample_free(&s->avr);
- }
-
- if (inlink->channel_layout == outlink->channel_layout &&
- inlink->sample_rate == outlink->sample_rate &&
- (inlink->format == outlink->format ||
- (av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
- av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
- av_get_planar_sample_fmt(inlink->format) ==
- av_get_planar_sample_fmt(outlink->format))))
- return 0;
-
- if (!(s->avr = avresample_alloc_context()))
- return AVERROR(ENOMEM);
-
- if (s->options) {
- int ret;
- AVDictionaryEntry *e = NULL;
- while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
- av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
-
- ret = av_opt_set_dict(s->avr, &s->options);
- if (ret < 0)
- return ret;
- }
-
- av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
- av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
- av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
- av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
- av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
- av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
-
- if ((ret = avresample_open(s->avr)) < 0)
- return ret;
-
- av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
- s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);
-
- if (s->resampling) {
- outlink->time_base = (AVRational){ 1, outlink->sample_rate };
- s->next_pts = AV_NOPTS_VALUE;
- s->next_in_pts = AV_NOPTS_VALUE;
- } else
- outlink->time_base = inlink->time_base;
-
- 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(ctx, AV_LOG_VERBOSE,
- "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
- av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
- av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
-
- return 0;
-}
-
-static int request_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- ResampleContext *s = ctx->priv;
- int ret = 0;
-
- s->got_output = 0;
- while (ret >= 0 && !s->got_output)
- ret = ff_request_frame(ctx->inputs[0]);
-
- /* flush the lavr delay buffer */
- if (ret == AVERROR_EOF && s->avr) {
- AVFrame *frame;
- int nb_samples = avresample_get_out_samples(s->avr, 0);
-
- if (!nb_samples)
- return ret;
-
- frame = ff_get_audio_buffer(outlink, nb_samples);
- if (!frame)
- return AVERROR(ENOMEM);
-
- ret = avresample_convert(s->avr, frame->extended_data,
- frame->linesize[0], nb_samples,
- NULL, 0, 0);
- if (ret <= 0) {
- av_frame_free(&frame);
- return (ret == 0) ? AVERROR_EOF : ret;
- }
-
- frame->nb_samples = ret;
- frame->pts = s->next_pts;
- return ff_filter_frame(outlink, frame);
- }
- return ret;
-}
-
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
-{
- AVFilterContext *ctx = inlink->dst;
- ResampleContext *s = ctx->priv;
- AVFilterLink *outlink = ctx->outputs[0];
- int ret;
-
- if (s->avr) {
- AVFrame *out;
- int delay, nb_samples;
-
- /* maximum possible samples lavr can output */
- delay = avresample_get_delay(s->avr);
- nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
-
- out = ff_get_audio_buffer(outlink, nb_samples);
- if (!out) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
-
- ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
- nb_samples, in->extended_data, in->linesize[0],
- in->nb_samples);
- if (ret <= 0) {
- av_frame_free(&out);
- if (ret < 0)
- goto fail;
- }
-
- av_assert0(!avresample_available(s->avr));
-
- if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
- if (in->pts == AV_NOPTS_VALUE) {
- av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
- "assuming 0.\n");
- s->next_pts = 0;
- } else
- s->next_pts = av_rescale_q(in->pts, inlink->time_base,
- outlink->time_base);
- }
-
- if (ret > 0) {
- out->nb_samples = ret;
-
- ret = av_frame_copy_props(out, in);
- if (ret < 0) {
- av_frame_free(&out);
- goto fail;
- }
-
- if (s->resampling) {
- out->sample_rate = outlink->sample_rate;
- /* Only convert in->pts if there is a discontinuous jump.
- This ensures that out->pts tracks the number of samples actually
- output by the resampler in the absence of such a jump.
- Otherwise, the rounding in av_rescale_q() and av_rescale()
- causes off-by-1 errors. */
- if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
- out->pts = av_rescale_q(in->pts, inlink->time_base,
- outlink->time_base) -
- av_rescale(delay, outlink->sample_rate,
- inlink->sample_rate);
- } else
- out->pts = s->next_pts;
-
- s->next_pts = out->pts + out->nb_samples;
- s->next_in_pts = in->pts + in->nb_samples;
- } else
- out->pts = in->pts;
-
- ret = ff_filter_frame(outlink, out);
- s->got_output = 1;
- }
-
-fail:
- av_frame_free(&in);
- } else {
- in->format = outlink->format;
- ret = ff_filter_frame(outlink, in);
- s->got_output = 1;
- }
-
- return ret;
-}
-
-#if FF_API_CHILD_CLASS_NEXT
-static const AVClass *resample_child_class_next(const AVClass *prev)
-{
- return prev ? NULL : avresample_get_class();
-}
-#endif
-
-static const AVClass *resample_child_class_iterate(void **iter)
-{
- const AVClass *c = *iter ? NULL : avresample_get_class();
- *iter = (void*)(uintptr_t)c;
- return c;
-}
-
-static void *resample_child_next(void *obj, void *prev)
-{
- ResampleContext *s = obj;
- return prev ? NULL : s->avr;
-}
-
-static const AVClass resample_class = {
- .class_name = "resample",
- .item_name = av_default_item_name,
- .version = LIBAVUTIL_VERSION_INT,
-#if FF_API_CHILD_CLASS_NEXT
- .child_class_next = resample_child_class_next,
-#endif
- .child_class_iterate = resample_child_class_iterate,
- .child_next = resample_child_next,
-};
-
-static const AVFilterPad avfilter_af_resample_inputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = filter_frame,
- },
- { NULL }
-};
-
-static const AVFilterPad avfilter_af_resample_outputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .config_props = config_output,
- .request_frame = request_frame
- },
- { NULL }
-};
-
-AVFilter ff_af_resample = {
- .name = "resample",
- .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
- .priv_size = sizeof(ResampleContext),
- .priv_class = &resample_class,
- .init_dict = init,
- .uninit = uninit,
- .query_formats = query_formats,
- .inputs = avfilter_af_resample_inputs,
- .outputs = avfilter_af_resample_outputs,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 7dbd1fb1dd..19c2acb63c 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -129,7 +129,6 @@ extern AVFilter ff_af_lv2;
extern AVFilter ff_af_mcompand;
extern AVFilter ff_af_pan;
extern AVFilter ff_af_replaygain;
-extern AVFilter ff_af_resample;
extern AVFilter ff_af_rubberband;
extern AVFilter ff_af_sidechaincompress;
extern AVFilter ff_af_sidechaingate;
diff --git a/libavresample/Makefile b/libavresample/Makefile
deleted file mode 100644
index 90f025a9f9..0000000000
--- a/libavresample/Makefile
+++ /dev/null
@@ -1,19 +0,0 @@
-NAME = avresample
-DESC = Libav audio resampling library
-
-HEADERS = avresample.h \
- version.h \
-
-OBJS = audio_convert.o \
- audio_data.o \
- audio_mix.o \
- audio_mix_matrix.o \
- dither.o \
- options.o \
- resample.o \
- utils.o \
-
-# Windows resource file
-SLIBOBJS-$(HAVE_GNU_WINDRES) += avresampleres.o
-
-TESTPROGS = avresample
diff --git a/libavresample/aarch64/Makefile b/libavresample/aarch64/Makefile
deleted file mode 100644
index f92699ef1a..0000000000
--- a/libavresample/aarch64/Makefile
+++ /dev/null
@@ -1,7 +0,0 @@
-OBJS += aarch64/audio_convert_init.o \
- aarch64/resample_init.o \
-
-OBJS-$(CONFIG_NEON_CLOBBER_TEST) += aarch64/neontest.o
-
-NEON-OBJS += aarch64/audio_convert_neon.o \
- aarch64/resample_neon.o \
diff --git a/libavresample/aarch64/asm-offsets.h b/libavresample/aarch64/asm-offsets.h
deleted file mode 100644
index 0b582446f6..0000000000
--- a/libavresample/aarch64/asm-offsets.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * 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 AVRESAMPLE_AARCH64_ASM_OFFSETS_H
-#define AVRESAMPLE_AARCH64_ASM_OFFSETS_H
-
-/* struct ResampleContext */
-#define FILTER_BANK 0x10
-#define FILTER_LENGTH 0x18
-#define PHASE_SHIFT 0x34
-#define PHASE_MASK (PHASE_SHIFT + 0x04) // loaded as pair
-
-#endif /* AVRESAMPLE_AARCH64_ASM_OFFSETS_H */
diff --git a/libavresample/aarch64/audio_convert_init.c b/libavresample/aarch64/audio_convert_init.c
deleted file mode 100644
index b5b0d1eee0..0000000000
--- a/libavresample/aarch64/audio_convert_init.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * 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 <stdint.h>
-
-#include "config.h"
-#include "libavutil/attributes.h"
-#include "libavutil/cpu.h"
-#include "libavutil/aarch64/cpu.h"
-#include "libavutil/samplefmt.h"
-#include "libavresample/audio_convert.h"
-
-void ff_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len);
-void ff_conv_fltp_to_s16_neon(int16_t *dst, float *const *src,
- int len, int channels);
-void ff_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src,
- int len, int channels);
-
-av_cold void ff_audio_convert_init_aarch64(AudioConvert *ac)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (have_neon(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
- 0, 16, 8, "NEON",
- ff_conv_flt_to_s16_neon);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 2, 16, 8, "NEON",
- ff_conv_fltp_to_s16_2ch_neon);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 0, 16, 8, "NEON",
- ff_conv_fltp_to_s16_neon);
- }
-}
diff --git a/libavresample/aarch64/audio_convert_neon.S b/libavresample/aarch64/audio_convert_neon.S
deleted file mode 100644
index e13e277e61..0000000000
--- a/libavresample/aarch64/audio_convert_neon.S
+++ /dev/null
@@ -1,363 +0,0 @@
-/*
- * Copyright (c) 2008 Mans Rullgard <mans at mansr.com>
- * Copyright (c) 2014 Janne Grunau <janne-libav at jannau.net>
- *
- * 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 "config.h"
-#include "libavutil/aarch64/asm.S"
-
-function ff_conv_flt_to_s16_neon, export=1
- subs x2, x2, #8
- ld1 {v0.4s}, [x1], #16
- fcvtzs v4.4s, v0.4s, #31
- ld1 {v1.4s}, [x1], #16
- fcvtzs v5.4s, v1.4s, #31
- b.eq 3f
- ands x12, x2, #~15
- b.eq 2f
-1: subs x12, x12, #16
- sqrshrn v4.4h, v4.4s, #16
- ld1 {v2.4s}, [x1], #16
- fcvtzs v6.4s, v2.4s, #31
- sqrshrn2 v4.8h, v5.4s, #16
- ld1 {v3.4s}, [x1], #16
- fcvtzs v7.4s, v3.4s, #31
- sqrshrn v6.4h, v6.4s, #16
- st1 {v4.8h}, [x0], #16
- sqrshrn2 v6.8h, v7.4s, #16
- ld1 {v0.4s}, [x1], #16
- fcvtzs v4.4s, v0.4s, #31
- ld1 {v1.4s}, [x1], #16
- fcvtzs v5.4s, v1.4s, #31
- st1 {v6.8h}, [x0], #16
- b.ne 1b
- ands x2, x2, #15
- b.eq 3f
-2: ld1 {v2.4s}, [x1], #16
- sqrshrn v4.4h, v4.4s, #16
- fcvtzs v6.4s, v2.4s, #31
- ld1 {v3.4s}, [x1], #16
- sqrshrn2 v4.8h, v5.4s, #16
- fcvtzs v7.4s, v3.4s, #31
- sqrshrn v6.4h, v6.4s, #16
- st1 {v4.8h}, [x0], #16
- sqrshrn2 v6.8h, v7.4s, #16
- st1 {v6.8h}, [x0]
- ret
-3: sqrshrn v4.4h, v4.4s, #16
- sqrshrn2 v4.8h, v5.4s, #16
- st1 {v4.8h}, [x0]
- ret
-endfunc
-
-function ff_conv_fltp_to_s16_2ch_neon, export=1
- ldp x4, x5, [x1]
- subs x2, x2, #8
- ld1 {v0.4s}, [x4], #16
- fcvtzs v4.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v5.4s, v1.4s, #31
- ld1 {v2.4s}, [x5], #16
- fcvtzs v6.4s, v2.4s, #31
- ld1 {v3.4s}, [x5], #16
- fcvtzs v7.4s, v3.4s, #31
- b.eq 3f
- ands x12, x2, #~15
- b.eq 2f
-1: subs x12, x12, #16
- ld1 {v16.4s}, [x4], #16
- fcvtzs v20.4s, v16.4s, #31
- sri v6.4s, v4.4s, #16
- ld1 {v17.4s}, [x4], #16
- fcvtzs v21.4s, v17.4s, #31
- ld1 {v18.4s}, [x5], #16
- fcvtzs v22.4s, v18.4s, #31
- ld1 {v19.4s}, [x5], #16
- sri v7.4s, v5.4s, #16
- st1 {v6.4s}, [x0], #16
- fcvtzs v23.4s, v19.4s, #31
- st1 {v7.4s}, [x0], #16
- sri v22.4s, v20.4s, #16
- ld1 {v0.4s}, [x4], #16
- sri v23.4s, v21.4s, #16
- st1 {v22.4s}, [x0], #16
- fcvtzs v4.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v5.4s, v1.4s, #31
- ld1 {v2.4s}, [x5], #16
- fcvtzs v6.4s, v2.4s, #31
- ld1 {v3.4s}, [x5], #16
- fcvtzs v7.4s, v3.4s, #31
- st1 {v23.4s}, [x0], #16
- b.ne 1b
- ands x2, x2, #15
- b.eq 3f
-2: sri v6.4s, v4.4s, #16
- ld1 {v0.4s}, [x4], #16
- fcvtzs v0.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v1.4s, v1.4s, #31
- ld1 {v2.4s}, [x5], #16
- fcvtzs v2.4s, v2.4s, #31
- sri v7.4s, v5.4s, #16
- ld1 {v3.4s}, [x5], #16
- fcvtzs v3.4s, v3.4s, #31
- sri v2.4s, v0.4s, #16
- st1 {v6.4s,v7.4s}, [x0], #32
- sri v3.4s, v1.4s, #16
- st1 {v2.4s,v3.4s}, [x0], #32
- ret
-3: sri v6.4s, v4.4s, #16
- sri v7.4s, v5.4s, #16
- st1 {v6.4s,v7.4s}, [x0]
- ret
-endfunc
-
-function ff_conv_fltp_to_s16_neon, export=1
- cmp w3, #2
- b.eq X(ff_conv_fltp_to_s16_2ch_neon)
- b.gt 1f
- ldr x1, [x1]
- b X(ff_conv_flt_to_s16_neon)
-1:
- cmp w3, #4
- lsl x12, x3, #1
- b.lt 4f
-
-5: // 4 channels
- ldp x4, x5, [x1], #16
- ldp x6, x7, [x1], #16
- mov w9, w2
- mov x8, x0
- ld1 {v4.4s}, [x4], #16
- fcvtzs v4.4s, v4.4s, #31
- ld1 {v5.4s}, [x5], #16
- fcvtzs v5.4s, v5.4s, #31
- ld1 {v6.4s}, [x6], #16
- fcvtzs v6.4s, v6.4s, #31
- ld1 {v7.4s}, [x7], #16
- fcvtzs v7.4s, v7.4s, #31
-6:
- subs w9, w9, #8
- ld1 {v0.4s}, [x4], #16
- fcvtzs v0.4s, v0.4s, #31
- sri v5.4s, v4.4s, #16
- ld1 {v1.4s}, [x5], #16
- fcvtzs v1.4s, v1.4s, #31
- sri v7.4s, v6.4s, #16
- ld1 {v2.4s}, [x6], #16
- fcvtzs v2.4s, v2.4s, #31
- zip1 v16.4s, v5.4s, v7.4s
- ld1 {v3.4s}, [x7], #16
- fcvtzs v3.4s, v3.4s, #31
- zip2 v17.4s, v5.4s, v7.4s
- st1 {v16.d}[0], [x8], x12
- sri v1.4s, v0.4s, #16
- st1 {v16.d}[1], [x8], x12
- sri v3.4s, v2.4s, #16
- st1 {v17.d}[0], [x8], x12
- zip1 v18.4s, v1.4s, v3.4s
- st1 {v17.d}[1], [x8], x12
- zip2 v19.4s, v1.4s, v3.4s
- b.eq 7f
- ld1 {v4.4s}, [x4], #16
- fcvtzs v4.4s, v4.4s, #31
- st1 {v18.d}[0], [x8], x12
- ld1 {v5.4s}, [x5], #16
- fcvtzs v5.4s, v5.4s, #31
- st1 {v18.d}[1], [x8], x12
- ld1 {v6.4s}, [x6], #16
- fcvtzs v6.4s, v6.4s, #31
- st1 {v19.d}[0], [x8], x12
- ld1 {v7.4s}, [x7], #16
- fcvtzs v7.4s, v7.4s, #31
- st1 {v19.d}[1], [x8], x12
- b 6b
-7:
- st1 {v18.d}[0], [x8], x12
- st1 {v18.d}[1], [x8], x12
- st1 {v19.d}[0], [x8], x12
- st1 {v19.d}[1], [x8], x12
- subs w3, w3, #4
- b.eq end
- cmp w3, #4
- add x0, x0, #8
- b.ge 5b
-
-4: // 2 channels
- cmp w3, #2
- b.lt 4f
- ldp x4, x5, [x1], #16
- mov w9, w2
- mov x8, x0
- tst w9, #8
- ld1 {v4.4s}, [x4], #16
- fcvtzs v4.4s, v4.4s, #31
- ld1 {v5.4s}, [x5], #16
- fcvtzs v5.4s, v5.4s, #31
- ld1 {v6.4s}, [x4], #16
- fcvtzs v6.4s, v6.4s, #31
- ld1 {v7.4s}, [x5], #16
- fcvtzs v7.4s, v7.4s, #31
- b.eq 6f
- subs w9, w9, #8
- b.eq 7f
- sri v5.4s, v4.4s, #16
- ld1 {v4.4s}, [x4], #16
- fcvtzs v4.4s, v4.4s, #31
- st1 {v5.s}[0], [x8], x12
- sri v7.4s, v6.4s, #16
- st1 {v5.s}[1], [x8], x12
- ld1 {v6.4s}, [x4], #16
- fcvtzs v6.4s, v6.4s, #31
- st1 {v5.s}[2], [x8], x12
- st1 {v5.s}[3], [x8], x12
- st1 {v7.s}[0], [x8], x12
- st1 {v7.s}[1], [x8], x12
- ld1 {v5.4s}, [x5], #16
- fcvtzs v5.4s, v5.4s, #31
- st1 {v7.s}[2], [x8], x12
- st1 {v7.s}[3], [x8], x12
- ld1 {v7.4s}, [x5], #16
- fcvtzs v7.4s, v7.4s, #31
-6:
- subs w9, w9, #16
- ld1 {v0.4s}, [x4], #16
- sri v5.4s, v4.4s, #16
- fcvtzs v0.4s, v0.4s, #31
- ld1 {v1.4s}, [x5], #16
- sri v7.4s, v6.4s, #16
- st1 {v5.s}[0], [x8], x12
- st1 {v5.s}[1], [x8], x12
- fcvtzs v1.4s, v1.4s, #31
- st1 {v5.s}[2], [x8], x12
- st1 {v5.s}[3], [x8], x12
- ld1 {v2.4s}, [x4], #16
- st1 {v7.s}[0], [x8], x12
- fcvtzs v2.4s, v2.4s, #31
- st1 {v7.s}[1], [x8], x12
- ld1 {v3.4s}, [x5], #16
- st1 {v7.s}[2], [x8], x12
- fcvtzs v3.4s, v3.4s, #31
- st1 {v7.s}[3], [x8], x12
- sri v1.4s, v0.4s, #16
- sri v3.4s, v2.4s, #16
- b.eq 6f
- ld1 {v4.4s}, [x4], #16
- st1 {v1.s}[0], [x8], x12
- fcvtzs v4.4s, v4.4s, #31
- st1 {v1.s}[1], [x8], x12
- ld1 {v5.4s}, [x5], #16
- st1 {v1.s}[2], [x8], x12
- fcvtzs v5.4s, v5.4s, #31
- st1 {v1.s}[3], [x8], x12
- ld1 {v6.4s}, [x4], #16
- st1 {v3.s}[0], [x8], x12
- fcvtzs v6.4s, v6.4s, #31
- st1 {v3.s}[1], [x8], x12
- ld1 {v7.4s}, [x5], #16
- st1 {v3.s}[2], [x8], x12
- fcvtzs v7.4s, v7.4s, #31
- st1 {v3.s}[3], [x8], x12
- b.gt 6b
-6:
- st1 {v1.s}[0], [x8], x12
- st1 {v1.s}[1], [x8], x12
- st1 {v1.s}[2], [x8], x12
- st1 {v1.s}[3], [x8], x12
- st1 {v3.s}[0], [x8], x12
- st1 {v3.s}[1], [x8], x12
- st1 {v3.s}[2], [x8], x12
- st1 {v3.s}[3], [x8], x12
- b 8f
-7:
- sri v5.4s, v4.4s, #16
- sri v7.4s, v6.4s, #16
- st1 {v5.s}[0], [x8], x12
- st1 {v5.s}[1], [x8], x12
- st1 {v5.s}[2], [x8], x12
- st1 {v5.s}[3], [x8], x12
- st1 {v7.s}[0], [x8], x12
- st1 {v7.s}[1], [x8], x12
- st1 {v7.s}[2], [x8], x12
- st1 {v7.s}[3], [x8], x12
-8:
- subs w3, w3, #2
- add x0, x0, #4
- b.eq end
-
-4: // 1 channel
- ldr x4, [x1]
- tst w2, #8
- mov w9, w2
- mov x5, x0
- ld1 {v0.4s}, [x4], #16
- fcvtzs v0.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v1.4s, v1.4s, #31
- b.ne 8f
-6:
- subs w9, w9, #16
- ld1 {v2.4s}, [x4], #16
- fcvtzs v2.4s, v2.4s, #31
- ld1 {v3.4s}, [x4], #16
- fcvtzs v3.4s, v3.4s, #31
- st1 {v0.h}[1], [x5], x12
- st1 {v0.h}[3], [x5], x12
- st1 {v0.h}[5], [x5], x12
- st1 {v0.h}[7], [x5], x12
- st1 {v1.h}[1], [x5], x12
- st1 {v1.h}[3], [x5], x12
- st1 {v1.h}[5], [x5], x12
- st1 {v1.h}[7], [x5], x12
- b.eq 7f
- ld1 {v0.4s}, [x4], #16
- fcvtzs v0.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v1.4s, v1.4s, #31
-7:
- st1 {v2.h}[1], [x5], x12
- st1 {v2.h}[3], [x5], x12
- st1 {v2.h}[5], [x5], x12
- st1 {v2.h}[7], [x5], x12
- st1 {v3.h}[1], [x5], x12
- st1 {v3.h}[3], [x5], x12
- st1 {v3.h}[5], [x5], x12
- st1 {v3.h}[7], [x5], x12
- b.gt 6b
- ret
-8:
- subs w9, w9, #8
- st1 {v0.h}[1], [x5], x12
- st1 {v0.h}[3], [x5], x12
- st1 {v0.h}[5], [x5], x12
- st1 {v0.h}[7], [x5], x12
- st1 {v1.h}[1], [x5], x12
- st1 {v1.h}[3], [x5], x12
- st1 {v1.h}[5], [x5], x12
- st1 {v1.h}[7], [x5], x12
- b.eq end
- ld1 {v0.4s}, [x4], #16
- fcvtzs v0.4s, v0.4s, #31
- ld1 {v1.4s}, [x4], #16
- fcvtzs v1.4s, v1.4s, #31
- b 6b
-end:
- ret
-endfunc
diff --git a/libavresample/aarch64/neontest.c b/libavresample/aarch64/neontest.c
deleted file mode 100644
index e956ee6b0d..0000000000
--- a/libavresample/aarch64/neontest.c
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * check NEON registers for clobbers
- * Copyright (c) 2013 Martin Storsjo
- *
- * 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 "libavresample/avresample.h"
-#include "libavutil/aarch64/neontest.h"
-
-wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
- int out_plane_size, int out_samples, uint8_t **input,
- int in_plane_size, int in_samples))
-{
- testneonclobbers(avresample_convert, avr, output, out_plane_size,
- out_samples, input, in_plane_size, in_samples);
-}
diff --git a/libavresample/aarch64/resample_init.c b/libavresample/aarch64/resample_init.c
deleted file mode 100644
index e21c600286..0000000000
--- a/libavresample/aarch64/resample_init.c
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * 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 <stdint.h>
-
-#include "config.h"
-#include "libavutil/cpu.h"
-#include "libavutil/aarch64/cpu.h"
-#include "libavutil/internal.h"
-#include "libavutil/samplefmt.h"
-#include "libavresample/resample.h"
-
-#include "asm-offsets.h"
-
-AV_CHECK_OFFSET(struct ResampleContext, filter_bank, FILTER_BANK);
-AV_CHECK_OFFSET(struct ResampleContext, filter_length, FILTER_LENGTH);
-AV_CHECK_OFFSET(struct ResampleContext, phase_shift, PHASE_SHIFT);
-AV_CHECK_OFFSET(struct ResampleContext, phase_mask, PHASE_MASK);
-
-void ff_resample_one_dbl_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-void ff_resample_one_flt_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-void ff_resample_one_s16_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-void ff_resample_one_s32_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-
-av_cold void ff_audio_resample_init_aarch64(ResampleContext *c,
- enum AVSampleFormat sample_fmt)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (have_neon(cpu_flags)) {
- if (!c->linear) {
- switch (sample_fmt) {
- case AV_SAMPLE_FMT_DBLP:
- c->resample_one = ff_resample_one_dbl_neon;
- break;
- case AV_SAMPLE_FMT_FLTP:
- c->resample_one = ff_resample_one_flt_neon;
- break;
- case AV_SAMPLE_FMT_S16P:
- c->resample_one = ff_resample_one_s16_neon;
- break;
- case AV_SAMPLE_FMT_S32P:
- c->resample_one = ff_resample_one_s32_neon;
- break;
- }
- }
- }
-}
diff --git a/libavresample/aarch64/resample_neon.S b/libavresample/aarch64/resample_neon.S
deleted file mode 100644
index d3c2cbf561..0000000000
--- a/libavresample/aarch64/resample_neon.S
+++ /dev/null
@@ -1,233 +0,0 @@
-/*
- * Copyright (c) 2014 Janne Grunau <janne-libav at jannau.net>
- *
- * 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 "libavutil/aarch64/asm.S"
-#include "asm-offsets.h"
-
-.macro resample_one fmt, es=2
-.ifnc \fmt, dbl
- .macro M_MUL2 x:vararg
- .endm
- .macro M_MLA2 x:vararg
- .endm
-.endif
-function ff_resample_one_\fmt\()_neon, export=1
- sxtw x2, w2
- ldr x9, [x0, #FILTER_BANK]
- ldr w6, [x0, #FILTER_LENGTH]
- ldp w7, w8, [x0, #PHASE_SHIFT] // and phase_mask
- lsr x10, x4, x7 // sample_index
- and x4, x4, x8
- lsl x11, x6, #\es // filter_length * elem_size
- add x3, x3, x10, lsl #\es // src[sample_index]
- madd x9, x11, x4, x9 // filter
- cmp w6, #16
- b.lt 5f
-8: // remaining filter_length at least 16
- subs w6, w6, #16
- LOAD8 v4, v5, v6, v7, x3
- LOAD8 v16, v17, v18, v19, x9
- M_MUL v0, v4, v16, v1
- M_MUL2 v1, v6, v18
-7:
- LOAD8 v20, v21, v22, v23, x3
- M_MLA v0, v5, v17, v1
- M_MLA2 v1, v7, v19
- LOAD8 v24, v25, v26, v27, x9
- M_MLA v0, v20, v24, v1
- M_MLA2 v1, v22, v26
- b.eq 6f
- cmp w6, #16
- M_MLA v0, v21, v25, v1
- M_MLA2 v1, v23, v27
- b.lt 4f
- subs w6, w6, #16
- LOAD8 v4, v5, v6, v7, x3
- LOAD8 v16, v17, v18, v19, x9
- M_MLA v0, v4, v16, v1
- M_MLA2 v1, v6, v18
- b 7b
-6:
- M_MLA v0, v21, v25, v1
- M_MLA2 v1, v23, v27
- STORE_ONE 0, x1, x2, v1
- ret
-5:
- movi v0.16b, #0
- movi v1.16b, #0
-4: // remaining filter_length 1-15
- cmp w6, #4
- b.lt 2f
- subs w6, w6, #4
- LOAD4 v4, v5, x3
- LOAD4 v6, v7, x9
- M_MLA v0, v4, v6, v1
- M_MLA2 v1, v5, v7
- b.eq 0f
- b 4b
-2: // remaining filter_length 1-3
- cmp w6, #2
- b.lt 1f
- LOAD2 2, x3
- LOAD2 3, x9
- subs w6, w6, #2
- M_MLA v0, v2, v3
- b.eq 0f
-1: // remaining filter_length 1
- LOAD1 6, x3
- LOAD1 7, x9
- M_MLA v0, v6, v7
-0:
- STORE_ONE 0, x1, x2, v1
- ret
-endfunc
-
-.purgem LOAD1
-.purgem LOAD2
-.purgem LOAD4
-.purgem LOAD8
-.purgem M_MLA
-.purgem M_MLA2
-.purgem M_MUL
-.purgem M_MUL2
-.purgem STORE_ONE
-.endm
-
-
-.macro LOAD1 d1, addr
- ldr d\d1, [\addr], #8
-.endm
-.macro LOAD2 d1, addr
- ld1 {v\d1\().2d}, [\addr], #16
-.endm
-.macro LOAD4 d1, d2, addr
- ld1 {\d1\().2d,\d2\().2d}, [\addr], #32
-.endm
-.macro LOAD8 d1, d2, d3, d4, addr
- ld1 {\d1\().2d,\d2\().2d,\d3\().2d,\d4\().2d}, [\addr], #64
-.endm
-.macro M_MLA d, r0, r1, d2:vararg
- fmla \d\().2d, \r0\().2d, \r1\().2d
-.endm
-.macro M_MLA2 second:vararg
- M_MLA \second
-.endm
-.macro M_MUL d, r0, r1, d2:vararg
- fmul \d\().2d, \r0\().2d, \r1\().2d
-.endm
-.macro M_MUL2 second:vararg
- M_MUL \second
-.endm
-.macro STORE_ONE rn, addr, idx, d2
- fadd v\rn\().2d, v\rn\().2d, \d2\().2d
- faddp d\rn\(), v\rn\().2d
- str d\rn\(), [\addr, \idx, lsl #3]
-.endm
-
-resample_one dbl, 3
-
-
-.macro LOAD1 d1, addr
- ldr s\d1, [\addr], #4
-.endm
-.macro LOAD2 d1, addr
- ld1 {v\d1\().2s}, [\addr], #8
-.endm
-.macro LOAD4 d1, d2, addr
- ld1 {\d1\().4s}, [\addr], #16
-.endm
-.macro LOAD8 d1, d2, d3, d4, addr
- ld1 {\d1\().4s,\d2\().4s}, [\addr], #32
-.endm
-.macro M_MLA d, r0, r1, d2:vararg
- fmla \d\().4s, \r0\().4s, \r1\().4s
-.endm
-.macro M_MUL d, r0, r1, d2:vararg
- fmul \d\().4s, \r0\().4s, \r1\().4s
-.endm
-.macro STORE_ONE rn, addr, idx, d2
- faddp v\rn\().4s, v\rn\().4s, v\rn\().4s
- faddp s\rn\(), v\rn\().2s
- str s\rn\(), [\addr, \idx, lsl #2]
-.endm
-
-resample_one flt
-
-
-.macro LOAD1 d1, addr
- ldr h\d1, [\addr], #2
-.endm
-.macro LOAD2 d1, addr
- ldr s\d1, [\addr], #4
-.endm
-.macro LOAD4 d1, d2, addr
- ld1 {\d1\().4h}, [\addr], #8
-.endm
-.macro LOAD8 d1, d2, d3, d4, addr
- ld1 {\d1\().4h,\d2\().4h}, [\addr], #16
-.endm
-.macro M_MLA d, r0, r1, d2:vararg
- smlal \d\().4s, \r0\().4h, \r1\().4h
-.endm
-.macro M_MUL d, r0, r1, d2:vararg
- smull \d\().4s, \r0\().4h, \r1\().4h
-.endm
-.macro STORE_ONE rn, addr, idx, d2
- addp v\rn\().4s, v\rn\().4s, v\rn\().4s
- addp v\rn\().4s, v\rn\().4s, v\rn\().4s
- sqrshrn v\rn\().4h, v\rn\().4s, #15
- str h\rn\(), [\addr, \idx, lsl #1]
-.endm
-
-resample_one s16, 1
-
-
-.macro LOAD1 d1, addr
- ldr s\d1, [\addr], #4
-.endm
-.macro LOAD2 d1, addr
- ld1 {v\d1\().2s}, [\addr], #8
-.endm
-.macro LOAD4 d1, d2, addr
- ld1 {\d1\().4s}, [\addr], #16
-.endm
-.macro LOAD8 d1, d2, d3, d4, addr
- ld1 {\d1\().4s,\d2\().4s}, [\addr], #32
-.endm
-.macro M_MLA d1, r0, r1, d2:vararg
- smlal \d1\().2d, \r0\().2s, \r1\().2s
-.ifnb \d2
- smlal2 \d2\().2d, \r0\().4s, \r1\().4s
-.endif
-.endm
-.macro M_MUL d1, r0, r1, d2:vararg
- smull \d1\().2d, \r0\().2s, \r1\().2s
-.ifnb \d2
- smull2 \d2\().2d, \r0\().4s, \r1\().4s
-.endif
-.endm
-.macro STORE_ONE rn, addr, idx, d2
- add v\rn\().2d, v\rn\().2d, \d2\().2d
- addp d\rn\(), v\rn\().2d
- sqrshrn v\rn\().2s, v\rn\().2d, #30
- str s\rn\(), [\addr, \idx, lsl #2]
-.endm
-
-resample_one s32
diff --git a/libavresample/arm/Makefile b/libavresample/arm/Makefile
deleted file mode 100644
index 352d1a8c13..0000000000
--- a/libavresample/arm/Makefile
+++ /dev/null
@@ -1,7 +0,0 @@
-OBJS += arm/audio_convert_init.o \
- arm/resample_init.o
-
-OBJS-$(CONFIG_NEON_CLOBBER_TEST) += arm/neontest.o
-
-NEON-OBJS += arm/audio_convert_neon.o \
- arm/resample_neon.o
diff --git a/libavresample/arm/asm-offsets.h b/libavresample/arm/asm-offsets.h
deleted file mode 100644
index 4d3d116dc0..0000000000
--- a/libavresample/arm/asm-offsets.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * 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 AVRESAMPLE_ARM_ASM_OFFSETS_H
-#define AVRESAMPLE_ARM_ASM_OFFSETS_H
-
-/* struct ResampleContext */
-#define FILTER_BANK 0x08
-#define FILTER_LENGTH 0x0c
-#define SRC_INCR 0x20
-#define PHASE_SHIFT 0x28
-#define PHASE_MASK (PHASE_SHIFT + 0x04)
-
-#endif /* AVRESAMPLE_ARM_ASM_OFFSETS_H */
diff --git a/libavresample/arm/audio_convert_init.c b/libavresample/arm/audio_convert_init.c
deleted file mode 100644
index 3d19a0e0e5..0000000000
--- a/libavresample/arm/audio_convert_init.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * 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 <stdint.h>
-
-#include "config.h"
-#include "libavutil/attributes.h"
-#include "libavutil/cpu.h"
-#include "libavutil/arm/cpu.h"
-#include "libavutil/samplefmt.h"
-#include "libavresample/audio_convert.h"
-
-void ff_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len);
-void ff_conv_fltp_to_s16_neon(int16_t *dst, float *const *src,
- int len, int channels);
-void ff_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src,
- int len, int channels);
-
-av_cold void ff_audio_convert_init_arm(AudioConvert *ac)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (have_neon(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
- 0, 16, 8, "NEON",
- ff_conv_flt_to_s16_neon);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 0, 16, 8, "NEON",
- ff_conv_fltp_to_s16_neon);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 2, 16, 8, "NEON",
- ff_conv_fltp_to_s16_2ch_neon);
- }
-}
diff --git a/libavresample/arm/audio_convert_neon.S b/libavresample/arm/audio_convert_neon.S
deleted file mode 100644
index a120e8793b..0000000000
--- a/libavresample/arm/audio_convert_neon.S
+++ /dev/null
@@ -1,363 +0,0 @@
-/*
- * Copyright (c) 2008 Mans Rullgard <mans at mansr.com>
- *
- * 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 "config.h"
-#include "libavutil/arm/asm.S"
-
-function ff_conv_flt_to_s16_neon, export=1
- subs r2, r2, #8
- vld1.32 {q0}, [r1,:128]!
- vcvt.s32.f32 q8, q0, #31
- vld1.32 {q1}, [r1,:128]!
- vcvt.s32.f32 q9, q1, #31
- beq 3f
- bics r12, r2, #15
- beq 2f
-1: subs r12, r12, #16
- vqrshrn.s32 d4, q8, #16
- vld1.32 {q0}, [r1,:128]!
- vcvt.s32.f32 q0, q0, #31
- vqrshrn.s32 d5, q9, #16
- vld1.32 {q1}, [r1,:128]!
- vcvt.s32.f32 q1, q1, #31
- vqrshrn.s32 d6, q0, #16
- vst1.16 {q2}, [r0,:128]!
- vqrshrn.s32 d7, q1, #16
- vld1.32 {q8}, [r1,:128]!
- vcvt.s32.f32 q8, q8, #31
- vld1.32 {q9}, [r1,:128]!
- vcvt.s32.f32 q9, q9, #31
- vst1.16 {q3}, [r0,:128]!
- bne 1b
- ands r2, r2, #15
- beq 3f
-2: vld1.32 {q0}, [r1,:128]!
- vqrshrn.s32 d4, q8, #16
- vcvt.s32.f32 q0, q0, #31
- vld1.32 {q1}, [r1,:128]!
- vqrshrn.s32 d5, q9, #16
- vcvt.s32.f32 q1, q1, #31
- vqrshrn.s32 d6, q0, #16
- vst1.16 {q2}, [r0,:128]!
- vqrshrn.s32 d7, q1, #16
- vst1.16 {q3}, [r0,:128]!
- bx lr
-3: vqrshrn.s32 d4, q8, #16
- vqrshrn.s32 d5, q9, #16
- vst1.16 {q2}, [r0,:128]!
- bx lr
-endfunc
-
-function ff_conv_fltp_to_s16_2ch_neon, export=1
- ldm r1, {r1, r3}
- subs r2, r2, #8
- vld1.32 {q0}, [r1,:128]!
- vcvt.s32.f32 q8, q0, #31
- vld1.32 {q1}, [r1,:128]!
- vcvt.s32.f32 q9, q1, #31
- vld1.32 {q10}, [r3,:128]!
- vcvt.s32.f32 q10, q10, #31
- vld1.32 {q11}, [r3,:128]!
- vcvt.s32.f32 q11, q11, #31
- beq 3f
- bics r12, r2, #15
- beq 2f
-1: subs r12, r12, #16
- vld1.32 {q0}, [r1,:128]!
- vcvt.s32.f32 q0, q0, #31
- vsri.32 q10, q8, #16
- vld1.32 {q1}, [r1,:128]!
- vcvt.s32.f32 q1, q1, #31
- vld1.32 {q12}, [r3,:128]!
- vcvt.s32.f32 q12, q12, #31
- vld1.32 {q13}, [r3,:128]!
- vsri.32 q11, q9, #16
- vst1.16 {q10}, [r0,:128]!
- vcvt.s32.f32 q13, q13, #31
- vst1.16 {q11}, [r0,:128]!
- vsri.32 q12, q0, #16
- vld1.32 {q8}, [r1,:128]!
- vsri.32 q13, q1, #16
- vst1.16 {q12}, [r0,:128]!
- vcvt.s32.f32 q8, q8, #31
- vld1.32 {q9}, [r1,:128]!
- vcvt.s32.f32 q9, q9, #31
- vld1.32 {q10}, [r3,:128]!
- vcvt.s32.f32 q10, q10, #31
- vld1.32 {q11}, [r3,:128]!
- vcvt.s32.f32 q11, q11, #31
- vst1.16 {q13}, [r0,:128]!
- bne 1b
- ands r2, r2, #15
- beq 3f
-2: vsri.32 q10, q8, #16
- vld1.32 {q0}, [r1,:128]!
- vcvt.s32.f32 q0, q0, #31
- vld1.32 {q1}, [r1,:128]!
- vcvt.s32.f32 q1, q1, #31
- vld1.32 {q12}, [r3,:128]!
- vcvt.s32.f32 q12, q12, #31
- vsri.32 q11, q9, #16
- vld1.32 {q13}, [r3,:128]!
- vcvt.s32.f32 q13, q13, #31
- vst1.16 {q10}, [r0,:128]!
- vsri.32 q12, q0, #16
- vst1.16 {q11}, [r0,:128]!
- vsri.32 q13, q1, #16
- vst1.16 {q12-q13},[r0,:128]!
- bx lr
-3: vsri.32 q10, q8, #16
- vsri.32 q11, q9, #16
- vst1.16 {q10-q11},[r0,:128]!
- bx lr
-endfunc
-
-function ff_conv_fltp_to_s16_neon, export=1
- cmp r3, #2
- itt lt
- ldrlt r1, [r1]
- blt X(ff_conv_flt_to_s16_neon)
- beq X(ff_conv_fltp_to_s16_2ch_neon)
-
- push {r4-r8, lr}
- cmp r3, #4
- lsl r12, r3, #1
- blt 4f
-
- @ 4 channels
-5: ldm r1!, {r4-r7}
- mov lr, r2
- mov r8, r0
- vld1.32 {q8}, [r4,:128]!
- vcvt.s32.f32 q8, q8, #31
- vld1.32 {q9}, [r5,:128]!
- vcvt.s32.f32 q9, q9, #31
- vld1.32 {q10}, [r6,:128]!
- vcvt.s32.f32 q10, q10, #31
- vld1.32 {q11}, [r7,:128]!
- vcvt.s32.f32 q11, q11, #31
-6: subs lr, lr, #8
- vld1.32 {q0}, [r4,:128]!
- vcvt.s32.f32 q0, q0, #31
- vsri.32 q9, q8, #16
- vld1.32 {q1}, [r5,:128]!
- vcvt.s32.f32 q1, q1, #31
- vsri.32 q11, q10, #16
- vld1.32 {q2}, [r6,:128]!
- vcvt.s32.f32 q2, q2, #31
- vzip.32 d18, d22
- vld1.32 {q3}, [r7,:128]!
- vcvt.s32.f32 q3, q3, #31
- vzip.32 d19, d23
- vst1.16 {d18}, [r8], r12
- vsri.32 q1, q0, #16
- vst1.16 {d22}, [r8], r12
- vsri.32 q3, q2, #16
- vst1.16 {d19}, [r8], r12
- vzip.32 d2, d6
- vst1.16 {d23}, [r8], r12
- vzip.32 d3, d7
- beq 7f
- vld1.32 {q8}, [r4,:128]!
- vcvt.s32.f32 q8, q8, #31
- vst1.16 {d2}, [r8], r12
- vld1.32 {q9}, [r5,:128]!
- vcvt.s32.f32 q9, q9, #31
- vst1.16 {d6}, [r8], r12
- vld1.32 {q10}, [r6,:128]!
- vcvt.s32.f32 q10, q10, #31
- vst1.16 {d3}, [r8], r12
- vld1.32 {q11}, [r7,:128]!
- vcvt.s32.f32 q11, q11, #31
- vst1.16 {d7}, [r8], r12
- b 6b
-7: vst1.16 {d2}, [r8], r12
- vst1.16 {d6}, [r8], r12
- vst1.16 {d3}, [r8], r12
- vst1.16 {d7}, [r8], r12
- subs r3, r3, #4
- it eq
- popeq {r4-r8, pc}
- cmp r3, #4
- add r0, r0, #8
- bge 5b
-
- @ 2 channels
-4: cmp r3, #2
- blt 4f
- ldm r1!, {r4-r5}
- mov lr, r2
- mov r8, r0
- tst lr, #8
- vld1.32 {q8}, [r4,:128]!
- vcvt.s32.f32 q8, q8, #31
- vld1.32 {q9}, [r5,:128]!
- vcvt.s32.f32 q9, q9, #31
- vld1.32 {q10}, [r4,:128]!
- vcvt.s32.f32 q10, q10, #31
- vld1.32 {q11}, [r5,:128]!
- vcvt.s32.f32 q11, q11, #31
- beq 6f
- subs lr, lr, #8
- beq 7f
- vsri.32 d18, d16, #16
- vsri.32 d19, d17, #16
- vld1.32 {q8}, [r4,:128]!
- vcvt.s32.f32 q8, q8, #31
- vst1.32 {d18[0]}, [r8], r12
- vsri.32 d22, d20, #16
- vst1.32 {d18[1]}, [r8], r12
- vsri.32 d23, d21, #16
- vst1.32 {d19[0]}, [r8], r12
- vst1.32 {d19[1]}, [r8], r12
- vld1.32 {q9}, [r5,:128]!
- vcvt.s32.f32 q9, q9, #31
- vst1.32 {d22[0]}, [r8], r12
- vst1.32 {d22[1]}, [r8], r12
- vld1.32 {q10}, [r4,:128]!
- vcvt.s32.f32 q10, q10, #31
- vst1.32 {d23[0]}, [r8], r12
- vst1.32 {d23[1]}, [r8], r12
- vld1.32 {q11}, [r5,:128]!
- vcvt.s32.f32 q11, q11, #31
-6: subs lr, lr, #16
- vld1.32 {q0}, [r4,:128]!
- vcvt.s32.f32 q0, q0, #31
- vsri.32 d18, d16, #16
- vld1.32 {q1}, [r5,:128]!
- vcvt.s32.f32 q1, q1, #31
- vsri.32 d19, d17, #16
- vld1.32 {q2}, [r4,:128]!
- vcvt.s32.f32 q2, q2, #31
- vld1.32 {q3}, [r5,:128]!
- vcvt.s32.f32 q3, q3, #31
- vst1.32 {d18[0]}, [r8], r12
- vsri.32 d22, d20, #16
- vst1.32 {d18[1]}, [r8], r12
- vsri.32 d23, d21, #16
- vst1.32 {d19[0]}, [r8], r12
- vsri.32 d2, d0, #16
- vst1.32 {d19[1]}, [r8], r12
- vsri.32 d3, d1, #16
- vst1.32 {d22[0]}, [r8], r12
- vsri.32 d6, d4, #16
- vst1.32 {d22[1]}, [r8], r12
- vsri.32 d7, d5, #16
- vst1.32 {d23[0]}, [r8], r12
- vst1.32 {d23[1]}, [r8], r12
- beq 6f
- vld1.32 {q8}, [r4,:128]!
- vcvt.s32.f32 q8, q8, #31
- vst1.32 {d2[0]}, [r8], r12
- vst1.32 {d2[1]}, [r8], r12
- vld1.32 {q9}, [r5,:128]!
- vcvt.s32.f32 q9, q9, #31
- vst1.32 {d3[0]}, [r8], r12
- vst1.32 {d3[1]}, [r8], r12
- vld1.32 {q10}, [r4,:128]!
- vcvt.s32.f32 q10, q10, #31
- vst1.32 {d6[0]}, [r8], r12
- vst1.32 {d6[1]}, [r8], r12
- vld1.32 {q11}, [r5,:128]!
- vcvt.s32.f32 q11, q11, #31
- vst1.32 {d7[0]}, [r8], r12
- vst1.32 {d7[1]}, [r8], r12
- bgt 6b
-6: vst1.32 {d2[0]}, [r8], r12
- vst1.32 {d2[1]}, [r8], r12
- vst1.32 {d3[0]}, [r8], r12
- vst1.32 {d3[1]}, [r8], r12
- vst1.32 {d6[0]}, [r8], r12
- vst1.32 {d6[1]}, [r8], r12
- vst1.32 {d7[0]}, [r8], r12
- vst1.32 {d7[1]}, [r8], r12
- b 8f
-7: vsri.32 d18, d16, #16
- vsri.32 d19, d17, #16
- vst1.32 {d18[0]}, [r8], r12
- vsri.32 d22, d20, #16
- vst1.32 {d18[1]}, [r8], r12
- vsri.32 d23, d21, #16
- vst1.32 {d19[0]}, [r8], r12
- vst1.32 {d19[1]}, [r8], r12
- vst1.32 {d22[0]}, [r8], r12
- vst1.32 {d22[1]}, [r8], r12
- vst1.32 {d23[0]}, [r8], r12
- vst1.32 {d23[1]}, [r8], r12
-8: subs r3, r3, #2
- add r0, r0, #4
- it eq
- popeq {r4-r8, pc}
-
- @ 1 channel
-4: ldr r4, [r1]
- tst r2, #8
- mov lr, r2
- mov r5, r0
- vld1.32 {q0}, [r4,:128]!
- vcvt.s32.f32 q0, q0, #31
- vld1.32 {q1}, [r4,:128]!
- vcvt.s32.f32 q1, q1, #31
- bne 8f
-6: subs lr, lr, #16
- vld1.32 {q2}, [r4,:128]!
- vcvt.s32.f32 q2, q2, #31
- vld1.32 {q3}, [r4,:128]!
- vcvt.s32.f32 q3, q3, #31
- vst1.16 {d0[1]}, [r5,:16], r12
- vst1.16 {d0[3]}, [r5,:16], r12
- vst1.16 {d1[1]}, [r5,:16], r12
- vst1.16 {d1[3]}, [r5,:16], r12
- vst1.16 {d2[1]}, [r5,:16], r12
- vst1.16 {d2[3]}, [r5,:16], r12
- vst1.16 {d3[1]}, [r5,:16], r12
- vst1.16 {d3[3]}, [r5,:16], r12
- beq 7f
- vld1.32 {q0}, [r4,:128]!
- vcvt.s32.f32 q0, q0, #31
- vld1.32 {q1}, [r4,:128]!
- vcvt.s32.f32 q1, q1, #31
-7: vst1.16 {d4[1]}, [r5,:16], r12
- vst1.16 {d4[3]}, [r5,:16], r12
- vst1.16 {d5[1]}, [r5,:16], r12
- vst1.16 {d5[3]}, [r5,:16], r12
- vst1.16 {d6[1]}, [r5,:16], r12
- vst1.16 {d6[3]}, [r5,:16], r12
- vst1.16 {d7[1]}, [r5,:16], r12
- vst1.16 {d7[3]}, [r5,:16], r12
- bgt 6b
- pop {r4-r8, pc}
-8: subs lr, lr, #8
- vst1.16 {d0[1]}, [r5,:16], r12
- vst1.16 {d0[3]}, [r5,:16], r12
- vst1.16 {d1[1]}, [r5,:16], r12
- vst1.16 {d1[3]}, [r5,:16], r12
- vst1.16 {d2[1]}, [r5,:16], r12
- vst1.16 {d2[3]}, [r5,:16], r12
- vst1.16 {d3[1]}, [r5,:16], r12
- vst1.16 {d3[3]}, [r5,:16], r12
- it eq
- popeq {r4-r8, pc}
- vld1.32 {q0}, [r4,:128]!
- vcvt.s32.f32 q0, q0, #31
- vld1.32 {q1}, [r4,:128]!
- vcvt.s32.f32 q1, q1, #31
- b 6b
-endfunc
diff --git a/libavresample/arm/neontest.c b/libavresample/arm/neontest.c
deleted file mode 100644
index 22afedbc60..0000000000
--- a/libavresample/arm/neontest.c
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * check NEON registers for clobbers
- * Copyright (c) 2013 Martin Storsjo
- *
- * 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 "libavresample/avresample.h"
-#include "libavutil/arm/neontest.h"
-
-wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
- int out_plane_size, int out_samples, uint8_t **input,
- int in_plane_size, int in_samples))
-{
- testneonclobbers(avresample_convert, avr, output, out_plane_size,
- out_samples, input, in_plane_size, in_samples);
-}
diff --git a/libavresample/arm/resample_init.c b/libavresample/arm/resample_init.c
deleted file mode 100644
index 10af09cb91..0000000000
--- a/libavresample/arm/resample_init.c
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2014 Peter Meerwald <pmeerw at pmeerw.net>
- *
- * 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 "config.h"
-
-#include "libavutil/cpu.h"
-#include "libavutil/arm/cpu.h"
-#include "libavutil/internal.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavresample/resample.h"
-
-#include "asm-offsets.h"
-
-AV_CHECK_OFFSET(struct ResampleContext, filter_bank, FILTER_BANK);
-AV_CHECK_OFFSET(struct ResampleContext, filter_length, FILTER_LENGTH);
-AV_CHECK_OFFSET(struct ResampleContext, src_incr, SRC_INCR);
-AV_CHECK_OFFSET(struct ResampleContext, phase_shift, PHASE_SHIFT);
-AV_CHECK_OFFSET(struct ResampleContext, phase_mask, PHASE_MASK);
-
-void ff_resample_one_flt_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-void ff_resample_one_s16_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-void ff_resample_one_s32_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-
-void ff_resample_linear_flt_neon(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
-
-av_cold void ff_audio_resample_init_arm(ResampleContext *c,
- enum AVSampleFormat sample_fmt)
-{
- int cpu_flags = av_get_cpu_flags();
- if (have_neon(cpu_flags)) {
- switch (sample_fmt) {
- case AV_SAMPLE_FMT_FLTP:
- if (c->linear)
- c->resample_one = ff_resample_linear_flt_neon;
- else
- c->resample_one = ff_resample_one_flt_neon;
- break;
- case AV_SAMPLE_FMT_S16P:
- if (!c->linear)
- c->resample_one = ff_resample_one_s16_neon;
- break;
- case AV_SAMPLE_FMT_S32P:
- if (!c->linear)
- c->resample_one = ff_resample_one_s32_neon;
- break;
- }
- }
-}
diff --git a/libavresample/arm/resample_neon.S b/libavresample/arm/resample_neon.S
deleted file mode 100644
index 7ee8497a5c..0000000000
--- a/libavresample/arm/resample_neon.S
+++ /dev/null
@@ -1,358 +0,0 @@
-/*
- * Copyright (c) 2014 Peter Meerwald <pmeerw at pmeerw.net>
- *
- * 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 "libavutil/arm/asm.S"
-
-#include "asm-offsets.h"
-
-.macro resample_one fmt, es=2
-function ff_resample_one_\fmt\()_neon, export=1
- push {r4, r5}
- add r1, r1, r2, lsl #\es
-
- ldr r2, [r0, #PHASE_SHIFT+4] /* phase_mask */
- ldr ip, [sp, #8] /* index */
- ldr r5, [r0, #FILTER_LENGTH]
- and r2, ip, r2 /* (index & phase_mask) */
- ldr r4, [r0, #PHASE_SHIFT]
- lsr r4, ip, r4 /* compute sample_index */
- mul r2, r2, r5
-
- ldr ip, [r0, #FILTER_BANK]
- add r3, r3, r4, lsl #\es /* &src[sample_index] */
-
- cmp r5, #8
- add r0, ip, r2, lsl #\es /* filter = &filter_bank[...] */
-
- blt 5f
-8:
- subs r5, r5, #8
- LOAD4
- MUL4
-7:
- LOAD4
- beq 6f
- cmp r5, #8
- MLA4
- blt 4f
- subs r5, r5, #8
- LOAD4
- MLA4
- b 7b
-6:
- MLA4
- STORE
- pop {r4, r5}
- bx lr
-5:
- INIT4
-4: /* remaining filter_length 1 to 7 */
- cmp r5, #4
- blt 2f
- subs r5, r5, #4
- LOAD4
- MLA4
- beq 0f
-2: /* remaining filter_length 1 to 3 */
- cmp r5, #2
- blt 1f
- subs r5, r5, #2
- LOAD2
- MLA2
- beq 0f
-1: /* remaining filter_length 1 */
- LOAD1
- MLA1
-0:
- STORE
- pop {r4, r5}
- bx lr
-endfunc
-
-.purgem LOAD1
-.purgem LOAD2
-.purgem LOAD4
-.purgem MLA1
-.purgem MLA2
-.purgem MLA4
-.purgem MUL4
-.purgem INIT4
-.purgem STORE
-.endm
-
-
-/* float32 */
-.macro LOAD1
- veor.32 d0, d0
- vld1.32 {d0[0]}, [r0]! /* load filter */
- vld1.32 {d4[0]}, [r3]! /* load src */
-.endm
-.macro LOAD2
- vld1.32 {d0}, [r0]! /* load filter */
- vld1.32 {d4}, [r3]! /* load src */
-.endm
-.macro LOAD4
- vld1.32 {d0,d1}, [r0]! /* load filter */
- vld1.32 {d4,d5}, [r3]! /* load src */
-.endm
-.macro MLA1
- vmla.f32 d16, d0, d4[0]
-.endm
-.macro MLA2
- vmla.f32 d16, d0, d4
-.endm
-.macro MLA4
- vmla.f32 d16, d0, d4
- vmla.f32 d17, d1, d5
-.endm
-.macro MUL4
- vmul.f32 d16, d0, d4
- vmul.f32 d17, d1, d5
-.endm
-.macro INIT4
- veor.f32 q8, q8
-.endm
-.macro STORE
- vpadd.f32 d16, d16, d17
- vpadd.f32 d16, d16, d16
- vst1.32 d16[0], [r1]
-.endm
-
-resample_one flt, 2
-
-
-/* s32 */
-.macro LOAD1
- veor.32 d0, d0
- vld1.32 {d0[0]}, [r0]! /* load filter */
- vld1.32 {d4[0]}, [r3]! /* load src */
-.endm
-.macro LOAD2
- vld1.32 {d0}, [r0]! /* load filter */
- vld1.32 {d4}, [r3]! /* load src */
-.endm
-.macro LOAD4
- vld1.32 {d0,d1}, [r0]! /* load filter */
- vld1.32 {d4,d5}, [r3]! /* load src */
-.endm
-.macro MLA1
- vmlal.s32 q8, d0, d4[0]
-.endm
-.macro MLA2
- vmlal.s32 q8, d0, d4
-.endm
-.macro MLA4
- vmlal.s32 q8, d0, d4
- vmlal.s32 q9, d1, d5
-.endm
-.macro MUL4
- vmull.s32 q8, d0, d4
- vmull.s32 q9, d1, d5
-.endm
-.macro INIT4
- veor.s64 q8, q8
- veor.s64 q9, q9
-.endm
-.macro STORE
- vadd.s64 q8, q8, q9
- vadd.s64 d16, d16, d17
- vqrshrn.s64 d16, q8, #30
- vst1.32 d16[0], [r1]
-.endm
-
-resample_one s32, 2
-
-
-/* s16 */
-.macro LOAD1
- veor.16 d0, d0
- vld1.16 {d0[0]}, [r0]! /* load filter */
- vld1.16 {d4[0]}, [r3]! /* load src */
-.endm
-.macro LOAD2
- veor.16 d0, d0
- vld1.32 {d0[0]}, [r0]! /* load filter */
- veor.16 d4, d4
- vld1.32 {d4[0]}, [r3]! /* load src */
-.endm
-.macro LOAD4
- vld1.16 {d0}, [r0]! /* load filter */
- vld1.16 {d4}, [r3]! /* load src */
-.endm
-.macro MLA1
- vmlal.s16 q8, d0, d4[0]
-.endm
-.macro MLA2
- vmlal.s16 q8, d0, d4
-.endm
-.macro MLA4
- vmlal.s16 q8, d0, d4
-.endm
-.macro MUL4
- vmull.s16 q8, d0, d4
-.endm
-.macro INIT4
- veor.s32 q8, q8
-.endm
-.macro STORE
- vpadd.s32 d16, d16, d17
- vpadd.s32 d16, d16, d16
- vqrshrn.s32 d16, q8, #15
- vst1.16 d16[0], [r1]
-.endm
-
-resample_one s16, 1
-
-
-.macro resample_linear fmt, es=2
-function ff_resample_linear_\fmt\()_neon, export=1
- push {r4, r5}
- add r1, r1, r2, lsl #\es
-
- ldr r2, [r0, #PHASE_SHIFT+4] /* phase_mask */
- ldr ip, [sp, #8] /* index */
- ldr r5, [r0, #FILTER_LENGTH]
- and r2, ip, r2 /* (index & phase_mask) */
- ldr r4, [r0, #PHASE_SHIFT]
- lsr r4, ip, r4 /* compute sample_index */
- mul r2, r2, r5
-
- ldr ip, [r0, #FILTER_BANK]
- add r3, r3, r4, lsl #\es /* &src[sample_index] */
-
- cmp r5, #8
- ldr r4, [r0, #SRC_INCR]
- add r0, ip, r2, lsl #\es /* filter = &filter_bank[...] */
- add r2, r0, r5, lsl #\es /* filter[... + c->filter_length] */
-
- blt 5f
-8:
- subs r5, r5, #8
- LOAD4
- MUL4
-7:
- LOAD4
- beq 6f
- cmp r5, #8
- MLA4
- blt 4f
- subs r5, r5, #8
- LOAD4
- MLA4
- b 7b
-6:
- MLA4
- STORE
- pop {r4, r5}
- bx lr
-5:
- INIT4
-4: /* remaining filter_length 1 to 7 */
- cmp r5, #4
- blt 2f
- subs r5, r5, #4
- LOAD4
- MLA4
- beq 0f
-2: /* remaining filter_length 1 to 3 */
- cmp r5, #2
- blt 1f
- subs r5, r5, #2
- LOAD2
- MLA2
- beq 0f
-1: /* remaining filter_length 1 */
- LOAD1
- MLA1
-0:
- STORE
- pop {r4, r5}
- bx lr
-endfunc
-
-.purgem LOAD1
-.purgem LOAD2
-.purgem LOAD4
-.purgem MLA1
-.purgem MLA2
-.purgem MLA4
-.purgem MUL4
-.purgem INIT4
-.purgem STORE
-.endm
-
-
-/* float32 linear */
-.macro LOAD1
- veor.32 d0, d0
- veor.32 d2, d2
- vld1.32 {d0[0]}, [r0]! /* load filter */
- vld1.32 {d2[0]}, [r2]! /* load filter */
- vld1.32 {d4[0]}, [r3]! /* load src */
-.endm
-.macro LOAD2
- vld1.32 {d0}, [r0]! /* load filter */
- vld1.32 {d2}, [r2]! /* load filter */
- vld1.32 {d4}, [r3]! /* load src */
-.endm
-.macro LOAD4
- vld1.32 {d0,d1}, [r0]! /* load filter */
- vld1.32 {d2,d3}, [r2]! /* load filter */
- vld1.32 {d4,d5}, [r3]! /* load src */
-.endm
-.macro MLA1
- vmla.f32 d18, d0, d4[0]
- vmla.f32 d16, d2, d4[0]
-.endm
-.macro MLA2
- vmla.f32 d18, d0, d4
- vmla.f32 d16, d2, d4
-.endm
-.macro MLA4
- vmla.f32 q9, q0, q2
- vmla.f32 q8, q1, q2
-.endm
-.macro MUL4
- vmul.f32 q9, q0, q2
- vmul.f32 q8, q1, q2
-.endm
-.macro INIT4
- veor.f32 q9, q9
- veor.f32 q8, q8
-.endm
-.macro STORE
- vldr s0, [sp, #12] /* frac */
- vmov s1, r4
- vcvt.f32.s32 d0, d0
-
- vsub.f32 q8, q8, q9 /* v2 - val */
- vpadd.f32 d18, d18, d19
- vpadd.f32 d16, d16, d17
- vpadd.f32 d2, d18, d18
- vpadd.f32 d1, d16, d16
-
- vmul.f32 s2, s2, s0 /* (v2 - val) * frac */
- vdiv.f32 s2, s2, s1 /* / c->src_incr */
- vadd.f32 s4, s4, s2
-
- vstr s4, [r1]
-.endm
-
-resample_linear flt, 2
diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c
deleted file mode 100644
index f2888cdd17..0000000000
--- a/libavresample/audio_convert.c
+++ /dev/null
@@ -1,416 +0,0 @@
-/*
- * Copyright (c) 2006 Michael Niedermayer <michaelni at gmx.at>
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-
-#include "config.h"
-#include "libavutil/common.h"
-#include "libavutil/libm.h"
-#include "libavutil/log.h"
-#include "libavutil/mem.h"
-#include "libavutil/samplefmt.h"
-#include "audio_convert.h"
-#include "audio_data.h"
-#include "dither.h"
-
-enum ConvFuncType {
- CONV_FUNC_TYPE_FLAT,
- CONV_FUNC_TYPE_INTERLEAVE,
- CONV_FUNC_TYPE_DEINTERLEAVE,
-};
-
-typedef void (conv_func_flat)(uint8_t *out, const uint8_t *in, int len);
-
-typedef void (conv_func_interleave)(uint8_t *out, uint8_t *const *in,
- int len, int channels);
-
-typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len,
- int channels);
-
-struct AudioConvert {
- AVAudioResampleContext *avr;
- DitherContext *dc;
- enum AVSampleFormat in_fmt;
- enum AVSampleFormat out_fmt;
- int apply_map;
- int channels;
- int planes;
- int ptr_align;
- int samples_align;
- int has_optimized_func;
- const char *func_descr;
- const char *func_descr_generic;
- enum ConvFuncType func_type;
- conv_func_flat *conv_flat;
- conv_func_flat *conv_flat_generic;
- conv_func_interleave *conv_interleave;
- conv_func_interleave *conv_interleave_generic;
- conv_func_deinterleave *conv_deinterleave;
- conv_func_deinterleave *conv_deinterleave_generic;
-};
-
-void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt, int channels,
- int ptr_align, int samples_align,
- const char *descr, void *conv)
-{
- int found = 0;
-
- switch (ac->func_type) {
- case CONV_FUNC_TYPE_FLAT:
- if (av_get_packed_sample_fmt(ac->in_fmt) == in_fmt &&
- av_get_packed_sample_fmt(ac->out_fmt) == out_fmt) {
- ac->conv_flat = conv;
- ac->func_descr = descr;
- ac->ptr_align = ptr_align;
- ac->samples_align = samples_align;
- if (ptr_align == 1 && samples_align == 1) {
- ac->conv_flat_generic = conv;
- ac->func_descr_generic = descr;
- } else {
- ac->has_optimized_func = 1;
- }
- found = 1;
- }
- break;
- case CONV_FUNC_TYPE_INTERLEAVE:
- if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
- (!channels || ac->channels == channels)) {
- ac->conv_interleave = conv;
- ac->func_descr = descr;
- ac->ptr_align = ptr_align;
- ac->samples_align = samples_align;
- if (ptr_align == 1 && samples_align == 1) {
- ac->conv_interleave_generic = conv;
- ac->func_descr_generic = descr;
- } else {
- ac->has_optimized_func = 1;
- }
- found = 1;
- }
- break;
- case CONV_FUNC_TYPE_DEINTERLEAVE:
- if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
- (!channels || ac->channels == channels)) {
- ac->conv_deinterleave = conv;
- ac->func_descr = descr;
- ac->ptr_align = ptr_align;
- ac->samples_align = samples_align;
- if (ptr_align == 1 && samples_align == 1) {
- ac->conv_deinterleave_generic = conv;
- ac->func_descr_generic = descr;
- } else {
- ac->has_optimized_func = 1;
- }
- found = 1;
- }
- break;
- }
- if (found) {
- av_log(ac->avr, AV_LOG_DEBUG, "audio_convert: found function: %-4s "
- "to %-4s (%s)\n", av_get_sample_fmt_name(ac->in_fmt),
- av_get_sample_fmt_name(ac->out_fmt), descr);
- }
-}
-
-#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
-
-#define CONV_LOOP(otype, expr) \
- do { \
- *(otype *)po = expr; \
- pi += is; \
- po += os; \
- } while (po < end); \
-
-#define CONV_FUNC_FLAT(ofmt, otype, ifmt, itype, expr) \
-static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t *in, \
- int len) \
-{ \
- int is = sizeof(itype); \
- int os = sizeof(otype); \
- const uint8_t *pi = in; \
- uint8_t *po = out; \
- uint8_t *end = out + os * len; \
- CONV_LOOP(otype, expr) \
-}
-
-#define CONV_FUNC_INTERLEAVE(ofmt, otype, ifmt, itype, expr) \
-static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t **in, \
- int len, int channels) \
-{ \
- int ch; \
- int out_bps = sizeof(otype); \
- int is = sizeof(itype); \
- int os = channels * out_bps; \
- for (ch = 0; ch < channels; ch++) { \
- const uint8_t *pi = in[ch]; \
- uint8_t *po = out + ch * out_bps; \
- uint8_t *end = po + os * len; \
- CONV_LOOP(otype, expr) \
- } \
-}
-
-#define CONV_FUNC_DEINTERLEAVE(ofmt, otype, ifmt, itype, expr) \
-static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t **out, const uint8_t *in, \
- int len, int channels) \
-{ \
- int ch; \
- int in_bps = sizeof(itype); \
- int is = channels * in_bps; \
- int os = sizeof(otype); \
- for (ch = 0; ch < channels; ch++) { \
- const uint8_t *pi = in + ch * in_bps; \
- uint8_t *po = out[ch]; \
- uint8_t *end = po + os * len; \
- CONV_LOOP(otype, expr) \
- } \
-}
-
-#define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr) \
-CONV_FUNC_FLAT( ofmt, otype, ifmt, itype, expr) \
-CONV_FUNC_INTERLEAVE( ofmt, otype, ifmt ## P, itype, expr) \
-CONV_FUNC_DEINTERLEAVE(ofmt ## P, otype, ifmt, itype, expr)
-
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_U8, uint8_t, *(const uint8_t *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 8)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 24)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0f / (1 << 7)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0 / (1 << 7)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t, (*(const int16_t *)pi >> 8) + 0x80)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi << 16)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0f / (1 << 15)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0 / (1 << 15)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t, (*(const int32_t *)pi >> 24) + 0x80)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi >> 16)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0f / (1U << 31)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0 / (1U << 31)))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8( lrintf(*(const float *)pi * (1 << 7)) + 0x80))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16( lrintf(*(const float *)pi * (1 << 15))))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *)pi * (1U << 31))))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8( lrint(*(const double *)pi * (1 << 7)) + 0x80))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16( lrint(*(const double *)pi * (1 << 15))))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *)pi * (1U << 31))))
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
-CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
-
-#define SET_CONV_FUNC_GROUP(ofmt, ifmt) \
-ff_audio_convert_set_func(ac, ofmt, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt)); \
-ff_audio_convert_set_func(ac, ofmt ## P, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt ## P, ifmt)); \
-ff_audio_convert_set_func(ac, ofmt, ifmt ## P, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt ## P));
-
-static void set_generic_function(AudioConvert *ac)
-{
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL)
- SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL)
-}
-
-void ff_audio_convert_free(AudioConvert **ac)
-{
- if (!*ac)
- return;
- ff_dither_free(&(*ac)->dc);
- av_freep(ac);
-}
-
-AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels, int sample_rate,
- int apply_map)
-{
- AudioConvert *ac;
- int in_planar, out_planar;
-
- ac = av_mallocz(sizeof(*ac));
- if (!ac)
- return NULL;
-
- ac->avr = avr;
- ac->out_fmt = out_fmt;
- ac->in_fmt = in_fmt;
- ac->channels = channels;
- ac->apply_map = apply_map;
-
- if (avr->dither_method != AV_RESAMPLE_DITHER_NONE &&
- av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
- av_get_bytes_per_sample(in_fmt) > 2) {
- ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate,
- apply_map);
- if (!ac->dc) {
- av_free(ac);
- return NULL;
- }
- return ac;
- }
-
- in_planar = ff_sample_fmt_is_planar(in_fmt, channels);
- out_planar = ff_sample_fmt_is_planar(out_fmt, channels);
-
- if (in_planar == out_planar) {
- ac->func_type = CONV_FUNC_TYPE_FLAT;
- ac->planes = in_planar ? ac->channels : 1;
- } else if (in_planar)
- ac->func_type = CONV_FUNC_TYPE_INTERLEAVE;
- else
- ac->func_type = CONV_FUNC_TYPE_DEINTERLEAVE;
-
- set_generic_function(ac);
-
- if (ARCH_AARCH64)
- ff_audio_convert_init_aarch64(ac);
- if (ARCH_ARM)
- ff_audio_convert_init_arm(ac);
- if (ARCH_X86)
- ff_audio_convert_init_x86(ac);
-
- return ac;
-}
-
-int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
-{
- int use_generic = 1;
- int len = in->nb_samples;
- int p;
-
- if (ac->dc) {
- /* dithered conversion */
- av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n",
- len, av_get_sample_fmt_name(ac->in_fmt),
- av_get_sample_fmt_name(ac->out_fmt));
-
- return ff_convert_dither(ac->dc, out, in);
- }
-
- /* determine whether to use the optimized function based on pointer and
- samples alignment in both the input and output */
- if (ac->has_optimized_func) {
- int ptr_align = FFMIN(in->ptr_align, out->ptr_align);
- int samples_align = FFMIN(in->samples_align, out->samples_align);
- int aligned_len = FFALIGN(len, ac->samples_align);
- if (!(ptr_align % ac->ptr_align) && samples_align >= aligned_len) {
- len = aligned_len;
- use_generic = 0;
- }
- }
- av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (%s)\n", len,
- av_get_sample_fmt_name(ac->in_fmt),
- av_get_sample_fmt_name(ac->out_fmt),
- use_generic ? ac->func_descr_generic : ac->func_descr);
-
- if (ac->apply_map) {
- ChannelMapInfo *map = &ac->avr->ch_map_info;
-
- if (!ff_sample_fmt_is_planar(ac->out_fmt, ac->channels)) {
- av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n");
- return AVERROR(EINVAL);
- }
-
- if (map->do_remap) {
- if (ff_sample_fmt_is_planar(ac->in_fmt, ac->channels)) {
- conv_func_flat *convert = use_generic ? ac->conv_flat_generic :
- ac->conv_flat;
-
- for (p = 0; p < ac->planes; p++)
- if (map->channel_map[p] >= 0)
- convert(out->data[p], in->data[map->channel_map[p]], len);
- } else {
- uint8_t *data[AVRESAMPLE_MAX_CHANNELS];
- conv_func_deinterleave *convert = use_generic ?
- ac->conv_deinterleave_generic :
- ac->conv_deinterleave;
-
- for (p = 0; p < ac->channels; p++)
- data[map->input_map[p]] = out->data[p];
-
- convert(data, in->data[0], len, ac->channels);
- }
- }
- if (map->do_copy || map->do_zero) {
- for (p = 0; p < ac->planes; p++) {
- if (map->channel_copy[p])
- memcpy(out->data[p], out->data[map->channel_copy[p]],
- len * out->stride);
- else if (map->channel_zero[p])
- av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt);
- }
- }
- } else {
- switch (ac->func_type) {
- case CONV_FUNC_TYPE_FLAT: {
- if (!in->is_planar)
- len *= in->channels;
- if (use_generic) {
- for (p = 0; p < ac->planes; p++)
- ac->conv_flat_generic(out->data[p], in->data[p], len);
- } else {
- for (p = 0; p < ac->planes; p++)
- ac->conv_flat(out->data[p], in->data[p], len);
- }
- break;
- }
- case CONV_FUNC_TYPE_INTERLEAVE:
- if (use_generic)
- ac->conv_interleave_generic(out->data[0], in->data, len,
- ac->channels);
- else
- ac->conv_interleave(out->data[0], in->data, len, ac->channels);
- break;
- case CONV_FUNC_TYPE_DEINTERLEAVE:
- if (use_generic)
- ac->conv_deinterleave_generic(out->data, in->data[0], len,
- ac->channels);
- else
- ac->conv_deinterleave(out->data, in->data[0], len,
- ac->channels);
- break;
- }
- }
-
- out->nb_samples = in->nb_samples;
- return 0;
-}
diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h
deleted file mode 100644
index df15442c18..0000000000
--- a/libavresample/audio_convert.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_AUDIO_CONVERT_H
-#define AVRESAMPLE_AUDIO_CONVERT_H
-
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-
-/**
- * Set conversion function if the parameters match.
- *
- * This compares the parameters of the conversion function to the parameters
- * in the AudioConvert context. If the parameters do not match, no changes are
- * made to the active functions. If the parameters do match and the alignment
- * is not constrained, the function is set as the generic conversion function.
- * If the parameters match and the alignment is constrained, the function is
- * set as the optimized conversion function.
- *
- * @param ac AudioConvert context
- * @param out_fmt output sample format
- * @param in_fmt input sample format
- * @param channels number of channels, or 0 for any number of channels
- * @param ptr_align buffer pointer alignment, in bytes
- * @param samples_align buffer size alignment, in samples
- * @param descr function type description (e.g. "C" or "SSE")
- * @param conv conversion function pointer
- */
-void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt, int channels,
- int ptr_align, int samples_align,
- const char *descr, void *conv);
-
-/**
- * Allocate and initialize AudioConvert context for sample format conversion.
- *
- * @param avr AVAudioResampleContext
- * @param out_fmt output sample format
- * @param in_fmt input sample format
- * @param channels number of channels
- * @param sample_rate sample rate (used for dithering)
- * @param apply_map apply channel map during conversion
- * @return newly-allocated AudioConvert context
- */
-AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels, int sample_rate,
- int apply_map);
-
-/**
- * Free AudioConvert.
- *
- * The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
- *
- * @param ac AudioConvert struct
- */
-void ff_audio_convert_free(AudioConvert **ac);
-
-/**
- * Convert audio data from one sample format to another.
- *
- * For each call, the alignment of the input and output AudioData buffers are
- * examined to determine whether to use the generic or optimized conversion
- * function (when available).
- *
- * The number of samples to convert is determined by in->nb_samples. The output
- * buffer must be large enough to handle this many samples. out->nb_samples is
- * set by this function before a successful return.
- *
- * @param ac AudioConvert context
- * @param out output audio data
- * @param in input audio data
- * @return 0 on success, negative AVERROR code on failure
- */
-int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in);
-
-/* arch-specific initialization functions */
-
-void ff_audio_convert_init_aarch64(AudioConvert *ac);
-void ff_audio_convert_init_arm(AudioConvert *ac);
-void ff_audio_convert_init_x86(AudioConvert *ac);
-
-#endif /* AVRESAMPLE_AUDIO_CONVERT_H */
diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c
deleted file mode 100644
index b54ead841a..0000000000
--- a/libavresample/audio_data.c
+++ /dev/null
@@ -1,381 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-#include <string.h>
-
-#include "libavutil/mem.h"
-#include "audio_data.h"
-
-static const AVClass audio_data_class = {
- .class_name = "AudioData",
- .item_name = av_default_item_name,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-/*
- * Calculate alignment for data pointers.
- */
-static void calc_ptr_alignment(AudioData *a)
-{
- int p;
- int min_align = 128;
-
- for (p = 0; p < a->planes; p++) {
- int cur_align = 128;
- while ((intptr_t)a->data[p] % cur_align)
- cur_align >>= 1;
- if (cur_align < min_align)
- min_align = cur_align;
- }
- a->ptr_align = min_align;
-}
-
-int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels)
-{
- if (channels == 1)
- return 1;
- else
- return av_sample_fmt_is_planar(sample_fmt);
-}
-
-int ff_audio_data_set_channels(AudioData *a, int channels)
-{
- if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
- channels > a->allocated_channels)
- return AVERROR(EINVAL);
-
- a->channels = channels;
- a->planes = a->is_planar ? channels : 1;
-
- calc_ptr_alignment(a);
-
- return 0;
-}
-
-int ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size,
- int channels, int nb_samples,
- enum AVSampleFormat sample_fmt, int read_only,
- const char *name)
-{
- int p;
-
- memset(a, 0, sizeof(*a));
- a->class = &audio_data_class;
-
- if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
- return AVERROR(EINVAL);
- }
-
- a->sample_size = av_get_bytes_per_sample(sample_fmt);
- if (!a->sample_size) {
- av_log(a, AV_LOG_ERROR, "invalid sample format\n");
- return AVERROR(EINVAL);
- }
- a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels);
- a->planes = a->is_planar ? channels : 1;
- a->stride = a->sample_size * (a->is_planar ? 1 : channels);
-
- for (p = 0; p < (a->is_planar ? channels : 1); p++) {
- if (!src[p]) {
- av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
- return AVERROR(EINVAL);
- }
- a->data[p] = src[p];
- }
- a->allocated_samples = nb_samples * !read_only;
- a->nb_samples = nb_samples;
- a->sample_fmt = sample_fmt;
- a->channels = channels;
- a->allocated_channels = channels;
- a->read_only = read_only;
- a->allow_realloc = 0;
- a->name = name ? name : "{no name}";
-
- calc_ptr_alignment(a);
- a->samples_align = plane_size / a->stride;
-
- return 0;
-}
-
-AudioData *ff_audio_data_alloc(int channels, int nb_samples,
- enum AVSampleFormat sample_fmt, const char *name)
-{
- AudioData *a;
- int ret;
-
- if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
- return NULL;
-
- a = av_mallocz(sizeof(*a));
- if (!a)
- return NULL;
-
- a->sample_size = av_get_bytes_per_sample(sample_fmt);
- if (!a->sample_size) {
- av_free(a);
- return NULL;
- }
- a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels);
- a->planes = a->is_planar ? channels : 1;
- a->stride = a->sample_size * (a->is_planar ? 1 : channels);
-
- a->class = &audio_data_class;
- a->sample_fmt = sample_fmt;
- a->channels = channels;
- a->allocated_channels = channels;
- a->read_only = 0;
- a->allow_realloc = 1;
- a->name = name ? name : "{no name}";
-
- if (nb_samples > 0) {
- ret = ff_audio_data_realloc(a, nb_samples);
- if (ret < 0) {
- av_free(a);
- return NULL;
- }
- return a;
- } else {
- calc_ptr_alignment(a);
- return a;
- }
-}
-
-int ff_audio_data_realloc(AudioData *a, int nb_samples)
-{
- int ret, new_buf_size, plane_size, p;
-
- /* check if buffer is already large enough */
- if (a->allocated_samples >= nb_samples)
- return 0;
-
- /* validate that the output is not read-only and realloc is allowed */
- if (a->read_only || !a->allow_realloc)
- return AVERROR(EINVAL);
-
- new_buf_size = av_samples_get_buffer_size(&plane_size,
- a->allocated_channels, nb_samples,
- a->sample_fmt, 0);
- if (new_buf_size < 0)
- return new_buf_size;
-
- /* if there is already data in the buffer and the sample format is planar,
- allocate a new buffer and copy the data, otherwise just realloc the
- internal buffer and set new data pointers */
- if (a->nb_samples > 0 && a->is_planar) {
- uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
-
- ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
- nb_samples, a->sample_fmt, 0);
- if (ret < 0)
- return ret;
-
- for (p = 0; p < a->planes; p++)
- memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
-
- av_freep(&a->buffer);
- memcpy(a->data, new_data, sizeof(new_data));
- a->buffer = a->data[0];
- } else {
- av_freep(&a->buffer);
- a->buffer = av_malloc(new_buf_size);
- if (!a->buffer)
- return AVERROR(ENOMEM);
- ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
- a->allocated_channels, nb_samples,
- a->sample_fmt, 0);
- if (ret < 0)
- return ret;
- }
- a->buffer_size = new_buf_size;
- a->allocated_samples = nb_samples;
-
- calc_ptr_alignment(a);
- a->samples_align = plane_size / a->stride;
-
- return 0;
-}
-
-void ff_audio_data_free(AudioData **a)
-{
- if (!*a)
- return;
- av_free((*a)->buffer);
- av_freep(a);
-}
-
-int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
-{
- int ret, p;
-
- /* validate input/output compatibility */
- if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
- return AVERROR(EINVAL);
-
- if (map && !src->is_planar) {
- av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
- return AVERROR(EINVAL);
- }
-
- /* if the input is empty, just empty the output */
- if (!src->nb_samples) {
- dst->nb_samples = 0;
- return 0;
- }
-
- /* reallocate output if necessary */
- ret = ff_audio_data_realloc(dst, src->nb_samples);
- if (ret < 0)
- return ret;
-
- /* copy data */
- if (map) {
- if (map->do_remap) {
- for (p = 0; p < src->planes; p++) {
- if (map->channel_map[p] >= 0)
- memcpy(dst->data[p], src->data[map->channel_map[p]],
- src->nb_samples * src->stride);
- }
- }
- if (map->do_copy || map->do_zero) {
- for (p = 0; p < src->planes; p++) {
- if (map->channel_copy[p])
- memcpy(dst->data[p], dst->data[map->channel_copy[p]],
- src->nb_samples * src->stride);
- else if (map->channel_zero[p])
- av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
- 1, dst->sample_fmt);
- }
- }
- } else {
- for (p = 0; p < src->planes; p++)
- memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
- }
-
- dst->nb_samples = src->nb_samples;
-
- return 0;
-}
-
-int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
- int src_offset, int nb_samples)
-{
- int ret, p, dst_offset2, dst_move_size;
-
- /* validate input/output compatibility */
- if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
- av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
- return AVERROR(EINVAL);
- }
-
- /* validate offsets are within the buffer bounds */
- if (dst_offset < 0 || dst_offset > dst->nb_samples ||
- src_offset < 0 || src_offset > src->nb_samples) {
- av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
- src_offset, dst_offset);
- return AVERROR(EINVAL);
- }
-
- /* check offsets and sizes to see if we can just do nothing and return */
- if (nb_samples > src->nb_samples - src_offset)
- nb_samples = src->nb_samples - src_offset;
- if (nb_samples <= 0)
- return 0;
-
- /* validate that the output is not read-only */
- if (dst->read_only) {
- av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
- return AVERROR(EINVAL);
- }
-
- /* reallocate output if necessary */
- ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
- if (ret < 0) {
- av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
- return ret;
- }
-
- dst_offset2 = dst_offset + nb_samples;
- dst_move_size = dst->nb_samples - dst_offset;
-
- for (p = 0; p < src->planes; p++) {
- if (dst_move_size > 0) {
- memmove(dst->data[p] + dst_offset2 * dst->stride,
- dst->data[p] + dst_offset * dst->stride,
- dst_move_size * dst->stride);
- }
- memcpy(dst->data[p] + dst_offset * dst->stride,
- src->data[p] + src_offset * src->stride,
- nb_samples * src->stride);
- }
- dst->nb_samples += nb_samples;
-
- return 0;
-}
-
-void ff_audio_data_drain(AudioData *a, int nb_samples)
-{
- if (a->nb_samples <= nb_samples) {
- /* drain the whole buffer */
- a->nb_samples = 0;
- } else {
- int p;
- int move_offset = a->stride * nb_samples;
- int move_size = a->stride * (a->nb_samples - nb_samples);
-
- for (p = 0; p < a->planes; p++)
- memmove(a->data[p], a->data[p] + move_offset, move_size);
-
- a->nb_samples -= nb_samples;
- }
-}
-
-int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
- int nb_samples)
-{
- uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
- int offset_size, p;
-
- if (offset >= a->nb_samples)
- return 0;
- offset_size = offset * a->stride;
- for (p = 0; p < a->planes; p++)
- offset_data[p] = a->data[p] + offset_size;
-
- return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
-}
-
-int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
-{
- int ret;
-
- if (a->read_only)
- return AVERROR(EINVAL);
-
- ret = ff_audio_data_realloc(a, nb_samples);
- if (ret < 0)
- return ret;
-
- ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
- if (ret >= 0)
- a->nb_samples = ret;
- return ret;
-}
diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h
deleted file mode 100644
index 1280307a95..0000000000
--- a/libavresample/audio_data.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_AUDIO_DATA_H
-#define AVRESAMPLE_AUDIO_DATA_H
-
-#include <stdint.h>
-
-#include "libavutil/audio_fifo.h"
-#include "libavutil/log.h"
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-#include "internal.h"
-
-int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels);
-
-/**
- * Audio buffer used for intermediate storage between conversion phases.
- */
-struct AudioData {
- const AVClass *class; /**< AVClass for logging */
- uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
- uint8_t *buffer; /**< data buffer */
- unsigned int buffer_size; /**< allocated buffer size */
- int allocated_samples; /**< number of samples the buffer can hold */
- int nb_samples; /**< current number of samples */
- enum AVSampleFormat sample_fmt; /**< sample format */
- int channels; /**< channel count */
- int allocated_channels; /**< allocated channel count */
- int is_planar; /**< sample format is planar */
- int planes; /**< number of data planes */
- int sample_size; /**< bytes per sample */
- int stride; /**< sample byte offset within a plane */
- int read_only; /**< data is read-only */
- int allow_realloc; /**< realloc is allowed */
- int ptr_align; /**< minimum data pointer alignment */
- int samples_align; /**< allocated samples alignment */
- const char *name; /**< name for debug logging */
-};
-
-int ff_audio_data_set_channels(AudioData *a, int channels);
-
-/**
- * Initialize AudioData using a given source.
- *
- * This does not allocate an internal buffer. It only sets the data pointers
- * and audio parameters.
- *
- * @param a AudioData struct
- * @param src source data pointers
- * @param plane_size plane size, in bytes.
- * This can be 0 if unknown, but that will lead to
- * optimized functions not being used in many cases,
- * which could slow down some conversions.
- * @param channels channel count
- * @param nb_samples number of samples in the source data
- * @param sample_fmt sample format
- * @param read_only indicates if buffer is read only or read/write
- * @param name name for debug logging (can be NULL)
- * @return 0 on success, negative AVERROR value on error
- */
-int ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size,
- int channels, int nb_samples,
- enum AVSampleFormat sample_fmt, int read_only,
- const char *name);
-
-/**
- * Allocate AudioData.
- *
- * This allocates an internal buffer and sets audio parameters.
- *
- * @param channels channel count
- * @param nb_samples number of samples to allocate space for
- * @param sample_fmt sample format
- * @param name name for debug logging (can be NULL)
- * @return newly allocated AudioData struct, or NULL on error
- */
-AudioData *ff_audio_data_alloc(int channels, int nb_samples,
- enum AVSampleFormat sample_fmt,
- const char *name);
-
-/**
- * Reallocate AudioData.
- *
- * The AudioData must have been previously allocated with ff_audio_data_alloc().
- *
- * @param a AudioData struct
- * @param nb_samples number of samples to allocate space for
- * @return 0 on success, negative AVERROR value on error
- */
-int ff_audio_data_realloc(AudioData *a, int nb_samples);
-
-/**
- * Free AudioData.
- *
- * The AudioData must have been previously allocated with ff_audio_data_alloc().
- *
- * @param a AudioData struct
- */
-void ff_audio_data_free(AudioData **a);
-
-/**
- * Copy data from one AudioData to another.
- *
- * @param out output AudioData
- * @param in input AudioData
- * @param map channel map, NULL if not remapping
- * @return 0 on success, negative AVERROR value on error
- */
-int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
-
-/**
- * Append data from one AudioData to the end of another.
- *
- * @param dst destination AudioData
- * @param dst_offset offset, in samples, to start writing, relative to the
- * start of dst
- * @param src source AudioData
- * @param src_offset offset, in samples, to start copying, relative to the
- * start of the src
- * @param nb_samples number of samples to copy
- * @return 0 on success, negative AVERROR value on error
- */
-int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
- int src_offset, int nb_samples);
-
-/**
- * Drain samples from the start of the AudioData.
- *
- * Remaining samples are shifted to the start of the AudioData.
- *
- * @param a AudioData struct
- * @param nb_samples number of samples to drain
- */
-void ff_audio_data_drain(AudioData *a, int nb_samples);
-
-/**
- * Add samples in AudioData to an AVAudioFifo.
- *
- * @param af Audio FIFO Buffer
- * @param a AudioData struct
- * @param offset number of samples to skip from the start of the data
- * @param nb_samples number of samples to add to the FIFO
- * @return number of samples actually added to the FIFO, or
- * negative AVERROR code on error
- */
-int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
- int nb_samples);
-
-/**
- * Read samples from an AVAudioFifo to AudioData.
- *
- * @param af Audio FIFO Buffer
- * @param a AudioData struct
- * @param nb_samples number of samples to read from the FIFO
- * @return number of samples actually read from the FIFO, or
- * negative AVERROR code on error
- */
-int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
-
-#endif /* AVRESAMPLE_AUDIO_DATA_H */
diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c
deleted file mode 100644
index 7ae0aeb74d..0000000000
--- a/libavresample/audio_mix.c
+++ /dev/null
@@ -1,742 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-
-#include "libavutil/common.h"
-#include "libavutil/libm.h"
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-#include "audio_mix.h"
-
-static const char * const coeff_type_names[] = { "q8", "q15", "flt" };
-
-struct AudioMix {
- AVAudioResampleContext *avr;
- enum AVSampleFormat fmt;
- enum AVMixCoeffType coeff_type;
- uint64_t in_layout;
- uint64_t out_layout;
- int in_channels;
- int out_channels;
-
- int ptr_align;
- int samples_align;
- int has_optimized_func;
- const char *func_descr;
- const char *func_descr_generic;
- mix_func *mix;
- mix_func *mix_generic;
-
- int in_matrix_channels;
- int out_matrix_channels;
- int output_zero[AVRESAMPLE_MAX_CHANNELS];
- int input_skip[AVRESAMPLE_MAX_CHANNELS];
- int output_skip[AVRESAMPLE_MAX_CHANNELS];
- int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS];
- int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
- float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
- void **matrix;
-};
-
-void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
- enum AVMixCoeffType coeff_type, int in_channels,
- int out_channels, int ptr_align, int samples_align,
- const char *descr, void *mix_func)
-{
- if (fmt == am->fmt && coeff_type == am->coeff_type &&
- ( in_channels == am->in_matrix_channels || in_channels == 0) &&
- (out_channels == am->out_matrix_channels || out_channels == 0)) {
- char chan_str[16];
- am->mix = mix_func;
- am->func_descr = descr;
- am->ptr_align = ptr_align;
- am->samples_align = samples_align;
- if (ptr_align == 1 && samples_align == 1) {
- am->mix_generic = mix_func;
- am->func_descr_generic = descr;
- } else {
- am->has_optimized_func = 1;
- }
- if (in_channels) {
- if (out_channels)
- snprintf(chan_str, sizeof(chan_str), "[%d to %d] ",
- in_channels, out_channels);
- else
- snprintf(chan_str, sizeof(chan_str), "[%d to any] ",
- in_channels);
- } else if (out_channels) {
- snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
- out_channels);
- } else {
- snprintf(chan_str, sizeof(chan_str), "[any to any] ");
- }
- av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
- "[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
- coeff_type_names[coeff_type], chan_str, descr);
- }
-}
-
-#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
-
-#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
-static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
- int len, int out_ch, int in_ch) \
-{ \
- int i, in, out; \
- stype temp[AVRESAMPLE_MAX_CHANNELS]; \
- for (i = 0; i < len; i++) { \
- for (out = 0; out < out_ch; out++) { \
- sumtype sum = 0; \
- for (in = 0; in < in_ch; in++) \
- sum += samples[in][i] * matrix[out][in]; \
- temp[out] = expr; \
- } \
- for (out = 0; out < out_ch; out++) \
- samples[out][i] = temp[out]; \
- } \
-}
-
-MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum)
-MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
-MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15))
-MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8))
-
-/* TODO: templatize the channel-specific C functions */
-
-static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
- int out_ch, int in_ch)
-{
- float *src0 = samples[0];
- float *src1 = samples[1];
- float *dst = src0;
- float m0 = matrix[0][0];
- float m1 = matrix[0][1];
-
- while (len > 4) {
- *dst++ = *src0++ * m0 + *src1++ * m1;
- *dst++ = *src0++ * m0 + *src1++ * m1;
- *dst++ = *src0++ * m0 + *src1++ * m1;
- *dst++ = *src0++ * m0 + *src1++ * m1;
- len -= 4;
- }
- while (len > 0) {
- *dst++ = *src0++ * m0 + *src1++ * m1;
- len--;
- }
-}
-
-static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len,
- int out_ch, int in_ch)
-{
- int16_t *src0 = samples[0];
- int16_t *src1 = samples[1];
- int16_t *dst = src0;
- float m0 = matrix[0][0];
- float m1 = matrix[0][1];
-
- while (len > 4) {
- *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
- *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
- *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
- *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
- len -= 4;
- }
- while (len > 0) {
- *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
- len--;
- }
-}
-
-static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len,
- int out_ch, int in_ch)
-{
- int16_t *src0 = samples[0];
- int16_t *src1 = samples[1];
- int16_t *dst = src0;
- int16_t m0 = matrix[0][0];
- int16_t m1 = matrix[0][1];
-
- while (len > 4) {
- *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
- *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
- *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
- *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
- len -= 4;
- }
- while (len > 0) {
- *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
- len--;
- }
-}
-
-static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
- int out_ch, int in_ch)
-{
- float v;
- float *dst0 = samples[0];
- float *dst1 = samples[1];
- float *src = dst0;
- float m0 = matrix[0][0];
- float m1 = matrix[1][0];
-
- while (len > 4) {
- v = *src++;
- *dst0++ = v * m0;
- *dst1++ = v * m1;
- v = *src++;
- *dst0++ = v * m0;
- *dst1++ = v * m1;
- v = *src++;
- *dst0++ = v * m0;
- *dst1++ = v * m1;
- v = *src++;
- *dst0++ = v * m0;
- *dst1++ = v * m1;
- len -= 4;
- }
- while (len > 0) {
- v = *src++;
- *dst0++ = v * m0;
- *dst1++ = v * m1;
- len--;
- }
-}
-
-static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len,
- int out_ch, int in_ch)
-{
- float v0, v1;
- float *src0 = samples[0];
- float *src1 = samples[1];
- float *src2 = samples[2];
- float *src3 = samples[3];
- float *src4 = samples[4];
- float *src5 = samples[5];
- float *dst0 = src0;
- float *dst1 = src1;
- float *m0 = matrix[0];
- float *m1 = matrix[1];
-
- while (len > 0) {
- v0 = *src0++;
- v1 = *src1++;
- *dst0++ = v0 * m0[0] +
- v1 * m0[1] +
- *src2 * m0[2] +
- *src3 * m0[3] +
- *src4 * m0[4] +
- *src5 * m0[5];
- *dst1++ = v0 * m1[0] +
- v1 * m1[1] +
- *src2++ * m1[2] +
- *src3++ * m1[3] +
- *src4++ * m1[4] +
- *src5++ * m1[5];
- len--;
- }
-}
-
-static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
- int out_ch, int in_ch)
-{
- float v0, v1;
- float *dst0 = samples[0];
- float *dst1 = samples[1];
- float *dst2 = samples[2];
- float *dst3 = samples[3];
- float *dst4 = samples[4];
- float *dst5 = samples[5];
- float *src0 = dst0;
- float *src1 = dst1;
-
- while (len > 0) {
- v0 = *src0++;
- v1 = *src1++;
- *dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
- *dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
- *dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
- *dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
- *dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
- *dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
- len--;
- }
-}
-
-static av_cold int mix_function_init(AudioMix *am)
-{
- am->func_descr = am->func_descr_generic = "n/a";
- am->mix = am->mix_generic = NULL;
-
- /* no need to set a mix function when we're skipping mixing */
- if (!am->in_matrix_channels || !am->out_matrix_channels)
- return 0;
-
- /* any-to-any C versions */
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT));
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT));
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15,
- 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15));
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
- 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q8));
-
- /* channel-specific C versions */
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c);
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
- 2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c);
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c);
-
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c);
-
- if (ARCH_X86)
- ff_audio_mix_init_x86(am);
-
- if (!am->mix) {
- av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
- "[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt),
- coeff_type_names[am->coeff_type], am->in_channels,
- am->out_channels);
- return AVERROR_PATCHWELCOME;
- }
- return 0;
-}
-
-AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr)
-{
- AudioMix *am;
- int ret;
-
- am = av_mallocz(sizeof(*am));
- if (!am)
- return NULL;
- am->avr = avr;
-
- if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
- av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
- "mixing: %s\n",
- av_get_sample_fmt_name(avr->internal_sample_fmt));
- goto error;
- }
-
- am->fmt = avr->internal_sample_fmt;
- am->coeff_type = avr->mix_coeff_type;
- am->in_layout = avr->in_channel_layout;
- am->out_layout = avr->out_channel_layout;
- am->in_channels = avr->in_channels;
- am->out_channels = avr->out_channels;
-
- /* build matrix if the user did not already set one */
- if (avr->mix_matrix) {
- ret = ff_audio_mix_set_matrix(am, avr->mix_matrix, avr->in_channels);
- if (ret < 0)
- goto error;
- av_freep(&avr->mix_matrix);
- } else {
- double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels *
- sizeof(*matrix_dbl));
- if (!matrix_dbl)
- goto error;
-
- ret = avresample_build_matrix(avr->in_channel_layout,
- avr->out_channel_layout,
- avr->center_mix_level,
- avr->surround_mix_level,
- avr->lfe_mix_level,
- avr->normalize_mix_level,
- matrix_dbl,
- avr->in_channels,
- avr->matrix_encoding);
- if (ret < 0) {
- av_free(matrix_dbl);
- goto error;
- }
-
- ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
- if (ret < 0) {
- av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n");
- av_free(matrix_dbl);
- goto error;
- }
-
- av_free(matrix_dbl);
- }
-
- return am;
-
-error:
- av_free(am);
- return NULL;
-}
-
-void ff_audio_mix_free(AudioMix **am_p)
-{
- AudioMix *am;
-
- if (!*am_p)
- return;
- am = *am_p;
-
- if (am->matrix) {
- av_free(am->matrix[0]);
- am->matrix = NULL;
- }
- memset(am->matrix_q8, 0, sizeof(am->matrix_q8 ));
- memset(am->matrix_q15, 0, sizeof(am->matrix_q15));
- memset(am->matrix_flt, 0, sizeof(am->matrix_flt));
-
- av_freep(am_p);
-}
-
-int ff_audio_mix(AudioMix *am, AudioData *src)
-{
- int use_generic = 1;
- int len = src->nb_samples;
- int i, j;
-
- /* determine whether to use the optimized function based on pointer and
- samples alignment in both the input and output */
- if (am->has_optimized_func) {
- int aligned_len = FFALIGN(len, am->samples_align);
- if (!(src->ptr_align % am->ptr_align) &&
- src->samples_align >= aligned_len) {
- len = aligned_len;
- use_generic = 0;
- }
- }
- av_log(am->avr, AV_LOG_TRACE, "audio_mix: %d samples - %d to %d channels (%s)\n",
- src->nb_samples, am->in_channels, am->out_channels,
- use_generic ? am->func_descr_generic : am->func_descr);
-
- if (am->in_matrix_channels && am->out_matrix_channels) {
- uint8_t **data;
- uint8_t *data0[AVRESAMPLE_MAX_CHANNELS] = { NULL };
-
- if (am->out_matrix_channels < am->out_channels ||
- am->in_matrix_channels < am->in_channels) {
- for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
- if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
- continue;
- data0[j++] = src->data[i];
- }
- data = data0;
- } else {
- data = src->data;
- }
-
- if (use_generic)
- am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
- am->in_matrix_channels);
- else
- am->mix(data, am->matrix, len, am->out_matrix_channels,
- am->in_matrix_channels);
- }
-
- if (am->out_matrix_channels < am->out_channels) {
- for (i = 0; i < am->out_channels; i++)
- if (am->output_zero[i])
- av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
- }
-
- ff_audio_data_set_channels(src, am->out_channels);
-
- return 0;
-}
-
-int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
-{
- int i, o, i0, o0;
-
- if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
- am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
- return AVERROR(EINVAL);
- }
-
-#define GET_MATRIX_CONVERT(suffix, scale) \
- if (!am->matrix_ ## suffix[0]) { \
- av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
- return AVERROR(EINVAL); \
- } \
- for (o = 0, o0 = 0; o < am->out_channels; o++) { \
- for (i = 0, i0 = 0; i < am->in_channels; i++) { \
- if (am->input_skip[i] || am->output_zero[o]) \
- matrix[o * stride + i] = 0.0; \
- else \
- matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
- (scale); \
- if (!am->input_skip[i]) \
- i0++; \
- } \
- if (!am->output_zero[o]) \
- o0++; \
- }
-
- switch (am->coeff_type) {
- case AV_MIX_COEFF_TYPE_Q8:
- GET_MATRIX_CONVERT(q8, 1.0 / 256.0);
- break;
- case AV_MIX_COEFF_TYPE_Q15:
- GET_MATRIX_CONVERT(q15, 1.0 / 32768.0);
- break;
- case AV_MIX_COEFF_TYPE_FLT:
- GET_MATRIX_CONVERT(flt, 1.0);
- break;
- default:
- av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
- return AVERROR(EINVAL);
- }
-
- return 0;
-}
-
-static void reduce_matrix(AudioMix *am, const double *matrix, int stride)
-{
- int i, o;
-
- memset(am->output_zero, 0, sizeof(am->output_zero));
- memset(am->input_skip, 0, sizeof(am->input_skip));
- memset(am->output_skip, 0, sizeof(am->output_skip));
-
- /* exclude output channels if they can be zeroed instead of mixed */
- for (o = 0; o < am->out_channels; o++) {
- int zero = 1;
-
- /* check if the output is always silent */
- for (i = 0; i < am->in_channels; i++) {
- if (matrix[o * stride + i] != 0.0) {
- zero = 0;
- break;
- }
- }
- /* check if the corresponding input channel makes a contribution to
- any output channel */
- if (o < am->in_channels) {
- for (i = 0; i < am->out_channels; i++) {
- if (matrix[i * stride + o] != 0.0) {
- zero = 0;
- break;
- }
- }
- }
- if (zero) {
- am->output_zero[o] = 1;
- am->out_matrix_channels--;
- if (o < am->in_channels)
- am->in_matrix_channels--;
- }
- }
- if (am->out_matrix_channels == 0 || am->in_matrix_channels == 0) {
- am->out_matrix_channels = 0;
- am->in_matrix_channels = 0;
- return;
- }
-
- /* skip input channels that contribute fully only to the corresponding
- output channel */
- for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) {
- int skip = 1;
-
- for (o = 0; o < am->out_channels; o++) {
- int i0;
- if ((o != i && matrix[o * stride + i] != 0.0) ||
- (o == i && matrix[o * stride + i] != 1.0)) {
- skip = 0;
- break;
- }
- /* if the input contributes fully to the output, also check that no
- other inputs contribute to this output */
- if (o == i) {
- for (i0 = 0; i0 < am->in_channels; i0++) {
- if (i0 != i && matrix[o * stride + i0] != 0.0) {
- skip = 0;
- break;
- }
- }
- }
- }
- if (skip) {
- am->input_skip[i] = 1;
- am->in_matrix_channels--;
- }
- }
- /* skip input channels that do not contribute to any output channel */
- for (; i < am->in_channels; i++) {
- int contrib = 0;
-
- for (o = 0; o < am->out_channels; o++) {
- if (matrix[o * stride + i] != 0.0) {
- contrib = 1;
- break;
- }
- }
- if (!contrib) {
- am->input_skip[i] = 1;
- am->in_matrix_channels--;
- }
- }
- if (am->in_matrix_channels == 0) {
- am->out_matrix_channels = 0;
- return;
- }
-
- /* skip output channels that only get full contribution from the
- corresponding input channel */
- for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) {
- int skip = 1;
- int o0;
-
- for (i = 0; i < am->in_channels; i++) {
- if ((o != i && matrix[o * stride + i] != 0.0) ||
- (o == i && matrix[o * stride + i] != 1.0)) {
- skip = 0;
- break;
- }
- }
- /* check if the corresponding input channel makes a contribution to
- any other output channel */
- i = o;
- for (o0 = 0; o0 < am->out_channels; o0++) {
- if (o0 != i && matrix[o0 * stride + i] != 0.0) {
- skip = 0;
- break;
- }
- }
- if (skip) {
- am->output_skip[o] = 1;
- am->out_matrix_channels--;
- }
- }
- if (am->out_matrix_channels == 0) {
- am->in_matrix_channels = 0;
- return;
- }
-}
-
-int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
-{
- int i, o, i0, o0, ret;
- char in_layout_name[128];
- char out_layout_name[128];
-
- if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
- am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n");
- return AVERROR(EINVAL);
- }
-
- if (am->matrix) {
- av_free(am->matrix[0]);
- am->matrix = NULL;
- }
-
- am->in_matrix_channels = am->in_channels;
- am->out_matrix_channels = am->out_channels;
-
- reduce_matrix(am, matrix, stride);
-
-#define CONVERT_MATRIX(type, expr) \
- am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
- am->in_matrix_channels * \
- sizeof(*am->matrix_## type[0])); \
- if (!am->matrix_## type[0]) \
- return AVERROR(ENOMEM); \
- for (o = 0, o0 = 0; o < am->out_channels; o++) { \
- if (am->output_zero[o] || am->output_skip[o]) \
- continue; \
- if (o0 > 0) \
- am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
- am->in_matrix_channels; \
- for (i = 0, i0 = 0; i < am->in_channels; i++) { \
- double v; \
- if (am->input_skip[i] || am->output_zero[i]) \
- continue; \
- v = matrix[o * stride + i]; \
- am->matrix_## type[o0][i0] = expr; \
- i0++; \
- } \
- o0++; \
- } \
- am->matrix = (void **)am->matrix_## type;
-
- if (am->in_matrix_channels && am->out_matrix_channels) {
- switch (am->coeff_type) {
- case AV_MIX_COEFF_TYPE_Q8:
- CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v)))
- break;
- case AV_MIX_COEFF_TYPE_Q15:
- CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v)))
- break;
- case AV_MIX_COEFF_TYPE_FLT:
- CONVERT_MATRIX(flt, v)
- break;
- default:
- av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
- return AVERROR(EINVAL);
- }
- }
-
- ret = mix_function_init(am);
- if (ret < 0)
- return ret;
-
- av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
- am->in_channels, am->in_layout);
- av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
- am->out_channels, am->out_layout);
- av_log(am->avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
- in_layout_name, out_layout_name);
- av_log(am->avr, AV_LOG_DEBUG, "matrix size: %d x %d\n",
- am->in_matrix_channels, am->out_matrix_channels);
- for (o = 0; o < am->out_channels; o++) {
- for (i = 0; i < am->in_channels; i++) {
- if (am->output_zero[o])
- av_log(am->avr, AV_LOG_DEBUG, " (ZERO)");
- else if (am->input_skip[i] || am->output_zero[i] || am->output_skip[o])
- av_log(am->avr, AV_LOG_DEBUG, " (SKIP)");
- else
- av_log(am->avr, AV_LOG_DEBUG, " %0.3f ",
- matrix[o * am->in_channels + i]);
- }
- av_log(am->avr, AV_LOG_DEBUG, "\n");
- }
-
- return 0;
-}
diff --git a/libavresample/audio_mix.h b/libavresample/audio_mix.h
deleted file mode 100644
index 0187d0f9e0..0000000000
--- a/libavresample/audio_mix.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_AUDIO_MIX_H
-#define AVRESAMPLE_AUDIO_MIX_H
-
-#include <stdint.h>
-
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-
-typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch,
- int in_ch);
-
-/**
- * Set mixing function if the parameters match.
- *
- * This compares the parameters of the mixing function to the parameters in the
- * AudioMix context. If the parameters do not match, no changes are made to the
- * active functions. If the parameters do match and the alignment is not
- * constrained, the function is set as the generic mixing function. If the
- * parameters match and the alignment is constrained, the function is set as
- * the optimized mixing function.
- *
- * @param am AudioMix context
- * @param fmt input/output sample format
- * @param coeff_type mixing coefficient type
- * @param in_channels number of input channels, or 0 for any number of channels
- * @param out_channels number of output channels, or 0 for any number of channels
- * @param ptr_align buffer pointer alignment, in bytes
- * @param samples_align buffer size alignment, in samples
- * @param descr function type description (e.g. "C" or "SSE")
- * @param mix_func mixing function pointer
- */
-void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
- enum AVMixCoeffType coeff_type, int in_channels,
- int out_channels, int ptr_align, int samples_align,
- const char *descr, void *mix_func);
-
-/**
- * Allocate and initialize an AudioMix context.
- *
- * The parameters in the AVAudioResampleContext are used to initialize the
- * AudioMix context.
- *
- * @param avr AVAudioResampleContext
- * @return newly-allocated AudioMix context.
- */
-AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr);
-
-/**
- * Free an AudioMix context.
- */
-void ff_audio_mix_free(AudioMix **am);
-
-/**
- * Apply channel mixing to audio data using the current mixing matrix.
- */
-int ff_audio_mix(AudioMix *am, AudioData *src);
-
-/**
- * Get the current mixing matrix.
- */
-int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride);
-
-/**
- * Set the current mixing matrix.
- */
-int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride);
-
-/* arch-specific initialization functions */
-
-void ff_audio_mix_init_x86(AudioMix *am);
-
-#endif /* AVRESAMPLE_AUDIO_MIX_H */
diff --git a/libavresample/audio_mix_matrix.c b/libavresample/audio_mix_matrix.c
deleted file mode 100644
index 5d92351a0e..0000000000
--- a/libavresample/audio_mix_matrix.c
+++ /dev/null
@@ -1,294 +0,0 @@
-/*
- * Copyright (C) 2011 Michael Niedermayer (michaelni at gmx.at)
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-
-#include "libavutil/common.h"
-#include "libavutil/libm.h"
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-#include "audio_mix.h"
-
-/* channel positions */
-#define FRONT_LEFT 0
-#define FRONT_RIGHT 1
-#define FRONT_CENTER 2
-#define LOW_FREQUENCY 3
-#define BACK_LEFT 4
-#define BACK_RIGHT 5
-#define FRONT_LEFT_OF_CENTER 6
-#define FRONT_RIGHT_OF_CENTER 7
-#define BACK_CENTER 8
-#define SIDE_LEFT 9
-#define SIDE_RIGHT 10
-#define TOP_CENTER 11
-#define TOP_FRONT_LEFT 12
-#define TOP_FRONT_CENTER 13
-#define TOP_FRONT_RIGHT 14
-#define TOP_BACK_LEFT 15
-#define TOP_BACK_CENTER 16
-#define TOP_BACK_RIGHT 17
-#define STEREO_LEFT 29
-#define STEREO_RIGHT 30
-#define WIDE_LEFT 31
-#define WIDE_RIGHT 32
-#define SURROUND_DIRECT_LEFT 33
-#define SURROUND_DIRECT_RIGHT 34
-#define LOW_FREQUENCY_2 35
-
-#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
-
-static av_always_inline int even(uint64_t layout)
-{
- return (!layout || !!(layout & (layout - 1)));
-}
-
-static int sane_layout(uint64_t layout)
-{
- /* check that there is at least 1 front speaker */
- if (!(layout & AV_CH_LAYOUT_SURROUND))
- return 0;
-
- /* check for left/right symmetry */
- if (!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT)) ||
- !even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT)) ||
- !even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)) ||
- !even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)) ||
- !even(layout & (AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT)) ||
- !even(layout & (AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT)) ||
- !even(layout & (AV_CH_STEREO_LEFT | AV_CH_STEREO_RIGHT)) ||
- !even(layout & (AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT)) ||
- !even(layout & (AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT)))
- return 0;
-
- return 1;
-}
-
-int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
- double center_mix_level, double surround_mix_level,
- double lfe_mix_level, int normalize,
- double *matrix_out, int stride,
- enum AVMatrixEncoding matrix_encoding)
-{
- int i, j, out_i, out_j;
- double matrix[64][64] = {{0}};
- int64_t unaccounted;
- double maxcoef = 0;
- int in_channels, out_channels;
-
- if ((out_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == AV_CH_LAYOUT_STEREO_DOWNMIX) {
- out_layout = AV_CH_LAYOUT_STEREO;
- }
-
- unaccounted = in_layout & ~out_layout;
-
- in_channels = av_get_channel_layout_nb_channels( in_layout);
- out_channels = av_get_channel_layout_nb_channels(out_layout);
-
- memset(matrix_out, 0, out_channels * stride * sizeof(*matrix_out));
-
- /* check if layouts are supported */
- if (!in_layout || in_channels > AVRESAMPLE_MAX_CHANNELS)
- return AVERROR(EINVAL);
- if (!out_layout || out_channels > AVRESAMPLE_MAX_CHANNELS)
- return AVERROR(EINVAL);
-
- /* check if layouts are unbalanced or abnormal */
- if (!sane_layout(in_layout) || !sane_layout(out_layout))
- return AVERROR_PATCHWELCOME;
-
- /* route matching input/output channels */
- for (i = 0; i < 64; i++) {
- if (in_layout & out_layout & (1ULL << i))
- matrix[i][i] = 1.0;
- }
-
- /* mix front center to front left/right */
- if (unaccounted & AV_CH_FRONT_CENTER) {
- if ((out_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) {
- if ((in_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) {
- matrix[FRONT_LEFT ][FRONT_CENTER] += center_mix_level;
- matrix[FRONT_RIGHT][FRONT_CENTER] += center_mix_level;
- } else {
- matrix[FRONT_LEFT ][FRONT_CENTER] += M_SQRT1_2;
- matrix[FRONT_RIGHT][FRONT_CENTER] += M_SQRT1_2;
- }
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix front left/right to center */
- if (unaccounted & AV_CH_LAYOUT_STEREO) {
- if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][FRONT_LEFT ] += M_SQRT1_2;
- matrix[FRONT_CENTER][FRONT_RIGHT] += M_SQRT1_2;
- /* mix left/right/center to center */
- if (in_layout & AV_CH_FRONT_CENTER)
- matrix[FRONT_CENTER][FRONT_CENTER] = center_mix_level * M_SQRT2;
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix back center to back, side, or front */
- if (unaccounted & AV_CH_BACK_CENTER) {
- if (out_layout & AV_CH_BACK_LEFT) {
- matrix[BACK_LEFT ][BACK_CENTER] += M_SQRT1_2;
- matrix[BACK_RIGHT][BACK_CENTER] += M_SQRT1_2;
- } else if (out_layout & AV_CH_SIDE_LEFT) {
- matrix[SIDE_LEFT ][BACK_CENTER] += M_SQRT1_2;
- matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2;
- } else if (out_layout & AV_CH_FRONT_LEFT) {
- if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY ||
- matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
- if (unaccounted & (AV_CH_BACK_LEFT | AV_CH_SIDE_LEFT)) {
- matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
- } else {
- matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level;
- matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level;
- }
- } else {
- matrix[FRONT_LEFT ][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
- }
- } else if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix back left/right to back center, side, or front */
- if (unaccounted & AV_CH_BACK_LEFT) {
- if (out_layout & AV_CH_BACK_CENTER) {
- matrix[BACK_CENTER][BACK_LEFT ] += M_SQRT1_2;
- matrix[BACK_CENTER][BACK_RIGHT] += M_SQRT1_2;
- } else if (out_layout & AV_CH_SIDE_LEFT) {
- /* if side channels do not exist in the input, just copy back
- channels to side channels, otherwise mix back into side */
- if (in_layout & AV_CH_SIDE_LEFT) {
- matrix[SIDE_LEFT ][BACK_LEFT ] += M_SQRT1_2;
- matrix[SIDE_RIGHT][BACK_RIGHT] += M_SQRT1_2;
- } else {
- matrix[SIDE_LEFT ][BACK_LEFT ] += 1.0;
- matrix[SIDE_RIGHT][BACK_RIGHT] += 1.0;
- }
- } else if (out_layout & AV_CH_FRONT_LEFT) {
- if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
- matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
- } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
- matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * SQRT3_2;
- matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * SQRT3_2;
- } else {
- matrix[FRONT_LEFT ][BACK_LEFT ] += surround_mix_level;
- matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level;
- }
- } else if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_CENTER][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix side left/right into back or front */
- if (unaccounted & AV_CH_SIDE_LEFT) {
- if (out_layout & AV_CH_BACK_LEFT) {
- /* if back channels do not exist in the input, just copy side
- channels to back channels, otherwise mix side into back */
- if (in_layout & AV_CH_BACK_LEFT) {
- matrix[BACK_LEFT ][SIDE_LEFT ] += M_SQRT1_2;
- matrix[BACK_RIGHT][SIDE_RIGHT] += M_SQRT1_2;
- } else {
- matrix[BACK_LEFT ][SIDE_LEFT ] += 1.0;
- matrix[BACK_RIGHT][SIDE_RIGHT] += 1.0;
- }
- } else if (out_layout & AV_CH_BACK_CENTER) {
- matrix[BACK_CENTER][SIDE_LEFT ] += M_SQRT1_2;
- matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2;
- } else if (out_layout & AV_CH_FRONT_LEFT) {
- if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) {
- matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;
- } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) {
- matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * SQRT3_2;
- matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * SQRT3_2;
- } else {
- matrix[FRONT_LEFT ][SIDE_LEFT ] += surround_mix_level;
- matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level;
- }
- } else if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
- matrix[FRONT_CENTER][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix left-of-center/right-of-center into front left/right or center */
- if (unaccounted & AV_CH_FRONT_LEFT_OF_CENTER) {
- if (out_layout & AV_CH_FRONT_LEFT) {
- matrix[FRONT_LEFT ][FRONT_LEFT_OF_CENTER ] += 1.0;
- matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER] += 1.0;
- } else if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][FRONT_LEFT_OF_CENTER ] += M_SQRT1_2;
- matrix[FRONT_CENTER][FRONT_RIGHT_OF_CENTER] += M_SQRT1_2;
- } else
- return AVERROR_PATCHWELCOME;
- }
- /* mix LFE into front left/right or center */
- if (unaccounted & AV_CH_LOW_FREQUENCY) {
- if (out_layout & AV_CH_FRONT_CENTER) {
- matrix[FRONT_CENTER][LOW_FREQUENCY] += lfe_mix_level;
- } else if (out_layout & AV_CH_FRONT_LEFT) {
- matrix[FRONT_LEFT ][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
- matrix[FRONT_RIGHT][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
- } else
- return AVERROR_PATCHWELCOME;
- }
-
- /* transfer internal matrix to output matrix and calculate maximum
- per-channel coefficient sum */
- for (out_i = i = 0; out_i < out_channels && i < 64; i++) {
- double sum = 0;
- for (out_j = j = 0; out_j < in_channels && j < 64; j++) {
- matrix_out[out_i * stride + out_j] = matrix[i][j];
- sum += fabs(matrix[i][j]);
- if (in_layout & (1ULL << j))
- out_j++;
- }
- maxcoef = FFMAX(maxcoef, sum);
- if (out_layout & (1ULL << i))
- out_i++;
- }
-
- /* normalize */
- if (normalize && maxcoef > 1.0) {
- for (i = 0; i < out_channels; i++)
- for (j = 0; j < in_channels; j++)
- matrix_out[i * stride + j] /= maxcoef;
- }
-
- return 0;
-}
diff --git a/libavresample/avresample.h b/libavresample/avresample.h
deleted file mode 100644
index 5ac9adb44b..0000000000
--- a/libavresample/avresample.h
+++ /dev/null
@@ -1,595 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_AVRESAMPLE_H
-#define AVRESAMPLE_AVRESAMPLE_H
-
-/**
- * @file
- * @ingroup lavr
- * external API header
- */
-
-/**
- * @defgroup lavr libavresample
- * @{
- *
- * Libavresample (lavr) is a library that handles audio resampling, sample
- * format conversion and mixing.
- *
- * Interaction with lavr is done through AVAudioResampleContext, which is
- * allocated with avresample_alloc_context(). It is opaque, so all parameters
- * must be set with the @ref avoptions API.
- *
- * For example the following code will setup conversion from planar float sample
- * format to interleaved signed 16-bit integer, downsampling from 48kHz to
- * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
- * matrix):
- * @code
- * AVAudioResampleContext *avr = avresample_alloc_context();
- * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
- * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
- * av_opt_set_int(avr, "in_sample_rate", 48000, 0);
- * av_opt_set_int(avr, "out_sample_rate", 44100, 0);
- * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
- * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
- * @endcode
- *
- * Once the context is initialized, it must be opened with avresample_open(). If
- * you need to change the conversion parameters, you must close the context with
- * avresample_close(), change the parameters as described above, then reopen it
- * again.
- *
- * The conversion itself is done by repeatedly calling avresample_convert().
- * Note that the samples may get buffered in two places in lavr. The first one
- * is the output FIFO, where the samples end up if the output buffer is not
- * large enough. The data stored in there may be retrieved at any time with
- * avresample_read(). The second place is the resampling delay buffer,
- * applicable only when resampling is done. The samples in it require more input
- * before they can be processed. Their current amount is returned by
- * avresample_get_delay(). At the end of conversion the resampling buffer can be
- * flushed by calling avresample_convert() with NULL input.
- *
- * The following code demonstrates the conversion loop assuming the parameters
- * from above and caller-defined functions get_input() and handle_output():
- * @code
- * uint8_t **input;
- * int in_linesize, in_samples;
- *
- * while (get_input(&input, &in_linesize, &in_samples)) {
- * uint8_t *output
- * int out_linesize;
- * int out_samples = avresample_get_out_samples(avr, in_samples);
- *
- * av_samples_alloc(&output, &out_linesize, 2, out_samples,
- * AV_SAMPLE_FMT_S16, 0);
- * out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
- * input, in_linesize, in_samples);
- * handle_output(output, out_linesize, out_samples);
- * av_freep(&output);
- * }
- * @endcode
- *
- * When the conversion is finished and the FIFOs are flushed if required, the
- * conversion context and everything associated with it must be freed with
- * avresample_free().
- */
-
-#include "libavutil/avutil.h"
-#include "libavutil/channel_layout.h"
-#include "libavutil/dict.h"
-#include "libavutil/frame.h"
-#include "libavutil/log.h"
-#include "libavutil/mathematics.h"
-
-#include "libavresample/version.h"
-
-#define AVRESAMPLE_MAX_CHANNELS 32
-
-typedef struct AVAudioResampleContext AVAudioResampleContext;
-
-/**
- * @deprecated use libswresample
- *
- * Mixing Coefficient Types */
-enum attribute_deprecated AVMixCoeffType {
- AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
- AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
- AV_MIX_COEFF_TYPE_FLT, /** floating-point */
- AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
-};
-
-/**
- * @deprecated use libswresample
- *
- * Resampling Filter Types */
-enum attribute_deprecated AVResampleFilterType {
- AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
- AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
- AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
-};
-
-/**
- * @deprecated use libswresample
- */
-enum attribute_deprecated AVResampleDitherMethod {
- AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
- AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
- AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
- AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
- AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
- AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
-};
-
-/**
- *
- * @deprecated use libswresample
- *
- * Return the LIBAVRESAMPLE_VERSION_INT constant.
- */
-attribute_deprecated
-unsigned avresample_version(void);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Return the libavresample build-time configuration.
- * @return configure string
- */
-attribute_deprecated
-const char *avresample_configuration(void);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Return the libavresample license.
- */
-attribute_deprecated
-const char *avresample_license(void);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Get the AVClass for AVAudioResampleContext.
- *
- * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
- * without allocating a context.
- *
- * @see av_opt_find().
- *
- * @return AVClass for AVAudioResampleContext
- */
-attribute_deprecated
-const AVClass *avresample_get_class(void);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Allocate AVAudioResampleContext and set options.
- *
- * @return allocated audio resample context, or NULL on failure
- */
-attribute_deprecated
-AVAudioResampleContext *avresample_alloc_context(void);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Initialize AVAudioResampleContext.
- * @note The context must be configured using the AVOption API.
- * @note The fields "in_channel_layout", "out_channel_layout",
- * "in_sample_rate", "out_sample_rate", "in_sample_fmt",
- * "out_sample_fmt" must be set.
- *
- * @see av_opt_set_int()
- * @see av_opt_set_dict()
- * @see av_get_default_channel_layout()
- *
- * @param avr audio resample context
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_open(AVAudioResampleContext *avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Check whether an AVAudioResampleContext is open or closed.
- *
- * @param avr AVAudioResampleContext to check
- * @return 1 if avr is open, 0 if avr is closed.
- */
-attribute_deprecated
-int avresample_is_open(AVAudioResampleContext *avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Close AVAudioResampleContext.
- *
- * This closes the context, but it does not change the parameters. The context
- * can be reopened with avresample_open(). It does, however, clear the output
- * FIFO and any remaining leftover samples in the resampling delay buffer. If
- * there was a custom matrix being used, that is also cleared.
- *
- * @see avresample_convert()
- * @see avresample_set_matrix()
- *
- * @param avr audio resample context
- */
-attribute_deprecated
-void avresample_close(AVAudioResampleContext *avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Free AVAudioResampleContext and associated AVOption values.
- *
- * This also calls avresample_close() before freeing.
- *
- * @param avr audio resample context
- */
-attribute_deprecated
-void avresample_free(AVAudioResampleContext **avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Generate a channel mixing matrix.
- *
- * This function is the one used internally by libavresample for building the
- * default mixing matrix. It is made public just as a utility function for
- * building custom matrices.
- *
- * @param in_layout input channel layout
- * @param out_layout output channel layout
- * @param center_mix_level mix level for the center channel
- * @param surround_mix_level mix level for the surround channel(s)
- * @param lfe_mix_level mix level for the low-frequency effects channel
- * @param normalize if 1, coefficients will be normalized to prevent
- * overflow. if 0, coefficients will not be
- * normalized.
- * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
- * the weight of input channel i in output channel o.
- * @param stride distance between adjacent input channels in the
- * matrix array
- * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
- double center_mix_level, double surround_mix_level,
- double lfe_mix_level, int normalize, double *matrix,
- int stride, enum AVMatrixEncoding matrix_encoding);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Get the current channel mixing matrix.
- *
- * If no custom matrix has been previously set or the AVAudioResampleContext is
- * not open, an error is returned.
- *
- * @param avr audio resample context
- * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
- * input channel i in output channel o.
- * @param stride distance between adjacent input channels in the matrix array
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
- int stride);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Set channel mixing matrix.
- *
- * Allows for setting a custom mixing matrix, overriding the default matrix
- * generated internally during avresample_open(). This function can be called
- * anytime on an allocated context, either before or after calling
- * avresample_open(), as long as the channel layouts have been set.
- * avresample_convert() always uses the current matrix.
- * Calling avresample_close() on the context will clear the current matrix.
- *
- * @see avresample_close()
- *
- * @param avr audio resample context
- * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
- * input channel i in output channel o.
- * @param stride distance between adjacent input channels in the matrix array
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
- int stride);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Set a customized input channel mapping.
- *
- * This function can only be called when the allocated context is not open.
- * Also, the input channel layout must have already been set.
- *
- * Calling avresample_close() on the context will clear the channel mapping.
- *
- * The map for each input channel specifies the channel index in the source to
- * use for that particular channel, or -1 to mute the channel. Source channels
- * can be duplicated by using the same index for multiple input channels.
- *
- * Examples:
- *
- * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
- * { 1, 2, 0, 5, 3, 4 }
- *
- * Muting the 3rd channel in 4-channel input:
- * { 0, 1, -1, 3 }
- *
- * Duplicating the left channel of stereo input:
- * { 0, 0 }
- *
- * @param avr audio resample context
- * @param channel_map customized input channel mapping
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_set_channel_mapping(AVAudioResampleContext *avr,
- const int *channel_map);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Set compensation for resampling.
- *
- * This can be called anytime after avresample_open(). If resampling is not
- * automatically enabled because of a sample rate conversion, the
- * "force_resampling" option must have been set to 1 when opening the context
- * in order to use resampling compensation.
- *
- * @param avr audio resample context
- * @param sample_delta compensation delta, in samples
- * @param compensation_distance compensation distance, in samples
- * @return 0 on success, negative AVERROR code on failure
- */
-attribute_deprecated
-int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
- int compensation_distance);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Provide the upper bound on the number of samples the configured
- * conversion would output.
- *
- * @param avr audio resample context
- * @param in_nb_samples number of input samples
- *
- * @return number of samples or AVERROR(EINVAL) if the value
- * would exceed INT_MAX
- */
-attribute_deprecated
-int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Convert input samples and write them to the output FIFO.
- *
- * The upper bound on the number of output samples can be obtained through
- * avresample_get_out_samples().
- *
- * The output data can be NULL or have fewer allocated samples than required.
- * In this case, any remaining samples not written to the output will be added
- * to an internal FIFO buffer, to be returned at the next call to this function
- * or to avresample_read().
- *
- * If converting sample rate, there may be data remaining in the internal
- * resampling delay buffer. avresample_get_delay() tells the number of remaining
- * samples. To get this data as output, call avresample_convert() with NULL
- * input.
- *
- * At the end of the conversion process, there may be data remaining in the
- * internal FIFO buffer. avresample_available() tells the number of remaining
- * samples. To get this data as output, either call avresample_convert() with
- * NULL input or call avresample_read().
- *
- * @see avresample_get_out_samples()
- * @see avresample_read()
- * @see avresample_get_delay()
- *
- * @param avr audio resample context
- * @param output output data pointers
- * @param out_plane_size output plane size, in bytes.
- * This can be 0 if unknown, but that will lead to
- * optimized functions not being used directly on the
- * output, which could slow down some conversions.
- * @param out_samples maximum number of samples that the output buffer can hold
- * @param input input data pointers
- * @param in_plane_size input plane size, in bytes
- * This can be 0 if unknown, but that will lead to
- * optimized functions not being used directly on the
- * input, which could slow down some conversions.
- * @param in_samples number of input samples to convert
- * @return number of samples written to the output buffer,
- * not including converted samples added to the internal
- * output FIFO
- */
-attribute_deprecated
-int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
- int out_plane_size, int out_samples,
- uint8_t * const *input, int in_plane_size,
- int in_samples);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Return the number of samples currently in the resampling delay buffer.
- *
- * When resampling, there may be a delay between the input and output. Any
- * unconverted samples in each call are stored internally in a delay buffer.
- * This function allows the user to determine the current number of samples in
- * the delay buffer, which can be useful for synchronization.
- *
- * @see avresample_convert()
- *
- * @param avr audio resample context
- * @return number of samples currently in the resampling delay buffer
- */
-attribute_deprecated
-int avresample_get_delay(AVAudioResampleContext *avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Return the number of available samples in the output FIFO.
- *
- * During conversion, if the user does not specify an output buffer or
- * specifies an output buffer that is smaller than what is needed, remaining
- * samples that are not written to the output are stored to an internal FIFO
- * buffer. The samples in the FIFO can be read with avresample_read() or
- * avresample_convert().
- *
- * @see avresample_read()
- * @see avresample_convert()
- *
- * @param avr audio resample context
- * @return number of samples available for reading
- */
-attribute_deprecated
-int avresample_available(AVAudioResampleContext *avr);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Read samples from the output FIFO.
- *
- * During conversion, if the user does not specify an output buffer or
- * specifies an output buffer that is smaller than what is needed, remaining
- * samples that are not written to the output are stored to an internal FIFO
- * buffer. This function can be used to read samples from that internal FIFO.
- *
- * @see avresample_available()
- * @see avresample_convert()
- *
- * @param avr audio resample context
- * @param output output data pointers. May be NULL, in which case
- * nb_samples of data is discarded from output FIFO.
- * @param nb_samples number of samples to read from the FIFO
- * @return the number of samples written to output
- */
-attribute_deprecated
-int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Convert the samples in the input AVFrame and write them to the output AVFrame.
- *
- * Input and output AVFrames must have channel_layout, sample_rate and format set.
- *
- * The upper bound on the number of output samples is obtained through
- * avresample_get_out_samples().
- *
- * If the output AVFrame does not have the data pointers allocated the nb_samples
- * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
- * is called to allocate the frame.
- *
- * The output AVFrame can be NULL or have fewer allocated samples than required.
- * In this case, any remaining samples not written to the output will be added
- * to an internal FIFO buffer, to be returned at the next call to this function
- * or to avresample_convert() or to avresample_read().
- *
- * If converting sample rate, there may be data remaining in the internal
- * resampling delay buffer. avresample_get_delay() tells the number of
- * remaining samples. To get this data as output, call this function or
- * avresample_convert() with NULL input.
- *
- * At the end of the conversion process, there may be data remaining in the
- * internal FIFO buffer. avresample_available() tells the number of remaining
- * samples. To get this data as output, either call this function or
- * avresample_convert() with NULL input or call avresample_read().
- *
- * If the AVAudioResampleContext configuration does not match the output and
- * input AVFrame settings the conversion does not take place and depending on
- * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
- * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
- *
- * @see avresample_get_out_samples()
- * @see avresample_available()
- * @see avresample_convert()
- * @see avresample_read()
- * @see avresample_get_delay()
- *
- * @param avr audio resample context
- * @param output output AVFrame
- * @param input input AVFrame
- * @return 0 on success, AVERROR on failure or nonmatching
- * configuration.
- */
-attribute_deprecated
-int avresample_convert_frame(AVAudioResampleContext *avr,
- AVFrame *output, AVFrame *input);
-
-/**
- *
- * @deprecated use libswresample
- *
- * Configure or reconfigure the AVAudioResampleContext using the information
- * provided by the AVFrames.
- *
- * The original resampling context is reset even on failure.
- * The function calls avresample_close() internally if the context is open.
- *
- * @see avresample_open();
- * @see avresample_close();
- *
- * @param avr audio resample context
- * @param out output AVFrame
- * @param in input AVFrame
- * @return 0 on success, AVERROR on failure.
- */
-attribute_deprecated
-int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
-
-/**
- * @}
- */
-
-#endif /* AVRESAMPLE_AVRESAMPLE_H */
diff --git a/libavresample/avresampleres.rc b/libavresample/avresampleres.rc
deleted file mode 100644
index e6d0d151e1..0000000000
--- a/libavresample/avresampleres.rc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Windows resource file for libavresample
- *
- * Copyright (C) 2012 James Almer
- * Copyright (C) 2013 Tiancheng "Timothy" Gu
- *
- * 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 <windows.h>
-#include "libavresample/version.h"
-#include "libavutil/ffversion.h"
-#include "config.h"
-
-1 VERSIONINFO
-FILEVERSION LIBAVRESAMPLE_VERSION_MAJOR, LIBAVRESAMPLE_VERSION_MINOR, LIBAVRESAMPLE_VERSION_MICRO, 0
-PRODUCTVERSION LIBAVRESAMPLE_VERSION_MAJOR, LIBAVRESAMPLE_VERSION_MINOR, LIBAVRESAMPLE_VERSION_MICRO, 0
-FILEFLAGSMASK VS_FFI_FILEFLAGSMASK
-FILEOS VOS_NT_WINDOWS32
-FILETYPE VFT_DLL
-{
- BLOCK "StringFileInfo"
- {
- BLOCK "040904B0"
- {
- VALUE "CompanyName", "FFmpeg Project"
- VALUE "FileDescription", "Libav audio resampling library"
- VALUE "FileVersion", AV_STRINGIFY(LIBAVRESAMPLE_VERSION)
- VALUE "InternalName", "libavresample"
- VALUE "LegalCopyright", "Copyright (C) 2000-" AV_STRINGIFY(CONFIG_THIS_YEAR) " FFmpeg Project"
- VALUE "OriginalFilename", "avresample" BUILDSUF "-" AV_STRINGIFY(LIBAVRESAMPLE_VERSION_MAJOR) SLIBSUF
- VALUE "ProductName", "FFmpeg"
- VALUE "ProductVersion", FFMPEG_VERSION
- }
- }
-
- BLOCK "VarFileInfo"
- {
- VALUE "Translation", 0x0409, 0x04B0
- }
-}
diff --git a/libavresample/dither.c b/libavresample/dither.c
deleted file mode 100644
index 2ae8d338be..0000000000
--- a/libavresample/dither.c
+++ /dev/null
@@ -1,440 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * Triangular with Noise Shaping is based on opusfile.
- * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors
- *
- * 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
- * Dithered Audio Sample Quantization
- *
- * Converts from dbl, flt, or s32 to s16 using dithering.
- */
-
-#include <math.h>
-#include <stdint.h>
-
-#include "libavutil/attributes.h"
-#include "libavutil/common.h"
-#include "libavutil/lfg.h"
-#include "libavutil/mem.h"
-#include "libavutil/samplefmt.h"
-#include "audio_convert.h"
-#include "dither.h"
-#include "internal.h"
-
-typedef struct DitherState {
- int mute;
- unsigned int seed;
- AVLFG lfg;
- float *noise_buf;
- int noise_buf_size;
- int noise_buf_ptr;
- float dither_a[4];
- float dither_b[4];
-} DitherState;
-
-struct DitherContext {
- DitherDSPContext ddsp;
- enum AVResampleDitherMethod method;
- int apply_map;
- ChannelMapInfo *ch_map_info;
-
- int mute_dither_threshold; // threshold for disabling dither
- int mute_reset_threshold; // threshold for resetting noise shaping
- const float *ns_coef_b; // noise shaping coeffs
- const float *ns_coef_a; // noise shaping coeffs
-
- int channels;
- DitherState *state; // dither states for each channel
-
- AudioData *flt_data; // input data in fltp
- AudioData *s16_data; // dithered output in s16p
- AudioConvert *ac_in; // converter for input to fltp
- AudioConvert *ac_out; // converter for s16p to s16 (if needed)
-
- void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
- int samples_align;
-};
-
-/* mute threshold, in seconds */
-#define MUTE_THRESHOLD_SEC 0.000333
-
-/* scale factor for 16-bit output.
- The signal is attenuated slightly to avoid clipping */
-#define S16_SCALE 32753.0f
-
-/* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */
-#define LFG_SCALE (1.0f / (2.0f * INT32_MAX))
-
-/* noise shaping coefficients */
-
-static const float ns_48_coef_b[4] = {
- 2.2374f, -0.7339f, -0.1251f, -0.6033f
-};
-
-static const float ns_48_coef_a[4] = {
- 0.9030f, 0.0116f, -0.5853f, -0.2571f
-};
-
-static const float ns_44_coef_b[4] = {
- 2.2061f, -0.4707f, -0.2534f, -0.6213f
-};
-
-static const float ns_44_coef_a[4] = {
- 1.0587f, 0.0676f, -0.6054f, -0.2738f
-};
-
-static void dither_int_to_float_rectangular_c(float *dst, int *src, int len)
-{
- int i;
- for (i = 0; i < len; i++)
- dst[i] = src[i] * LFG_SCALE;
-}
-
-static void dither_int_to_float_triangular_c(float *dst, int *src0, int len)
-{
- int i;
- int *src1 = src0 + len;
-
- for (i = 0; i < len; i++) {
- float r = src0[i] * LFG_SCALE;
- r += src1[i] * LFG_SCALE;
- dst[i] = r;
- }
-}
-
-static void quantize_c(int16_t *dst, const float *src, float *dither, int len)
-{
- int i;
- for (i = 0; i < len; i++)
- dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i]));
-}
-
-#define SQRT_1_6 0.40824829046386301723f
-
-static void dither_highpass_filter(float *src, int len)
-{
- int i;
-
- /* filter is from libswresample in FFmpeg */
- for (i = 0; i < len - 2; i++)
- src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6;
-}
-
-static int generate_dither_noise(DitherContext *c, DitherState *state,
- int min_samples)
-{
- int i;
- int nb_samples = FFALIGN(min_samples, 16) + 16;
- int buf_samples = nb_samples *
- (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2);
- unsigned int *noise_buf_ui;
-
- av_freep(&state->noise_buf);
- state->noise_buf_size = state->noise_buf_ptr = 0;
-
- state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf));
- if (!state->noise_buf)
- return AVERROR(ENOMEM);
- state->noise_buf_size = FFALIGN(min_samples, 16);
- noise_buf_ui = (unsigned int *)state->noise_buf;
-
- av_lfg_init(&state->lfg, state->seed);
- for (i = 0; i < buf_samples; i++)
- noise_buf_ui[i] = av_lfg_get(&state->lfg);
-
- c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples);
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP)
- dither_highpass_filter(state->noise_buf, nb_samples);
-
- return 0;
-}
-
-static void quantize_triangular_ns(DitherContext *c, DitherState *state,
- int16_t *dst, const float *src,
- int nb_samples)
-{
- int i, j;
- float *dither = &state->noise_buf[state->noise_buf_ptr];
-
- if (state->mute > c->mute_reset_threshold)
- memset(state->dither_a, 0, sizeof(state->dither_a));
-
- for (i = 0; i < nb_samples; i++) {
- float err = 0;
- float sample = src[i] * S16_SCALE;
-
- for (j = 0; j < 4; j++) {
- err += c->ns_coef_b[j] * state->dither_b[j] -
- c->ns_coef_a[j] * state->dither_a[j];
- }
- for (j = 3; j > 0; j--) {
- state->dither_a[j] = state->dither_a[j - 1];
- state->dither_b[j] = state->dither_b[j - 1];
- }
- state->dither_a[0] = err;
- sample -= err;
-
- if (state->mute > c->mute_dither_threshold) {
- dst[i] = av_clip_int16(lrintf(sample));
- state->dither_b[0] = 0;
- } else {
- dst[i] = av_clip_int16(lrintf(sample + dither[i]));
- state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f);
- }
-
- state->mute++;
- if (src[i])
- state->mute = 0;
- }
-}
-
-static int convert_samples(DitherContext *c, int16_t **dst, float * const *src,
- int channels, int nb_samples)
-{
- int ch, ret;
- int aligned_samples = FFALIGN(nb_samples, 16);
-
- for (ch = 0; ch < channels; ch++) {
- DitherState *state = &c->state[ch];
-
- if (state->noise_buf_size < aligned_samples) {
- ret = generate_dither_noise(c, state, nb_samples);
- if (ret < 0)
- return ret;
- } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) {
- state->noise_buf_ptr = 0;
- }
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples);
- } else {
- c->quantize(dst[ch], src[ch],
- &state->noise_buf[state->noise_buf_ptr],
- FFALIGN(nb_samples, c->samples_align));
- }
-
- state->noise_buf_ptr += aligned_samples;
- }
-
- return 0;
-}
-
-int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
-{
- int ret;
- AudioData *flt_data;
-
- /* output directly to dst if it is planar */
- if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
- c->s16_data = dst;
- else {
- /* make sure s16_data is large enough for the output */
- ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
- if (ret < 0)
- return ret;
- }
-
- if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
- /* make sure flt_data is large enough for the input */
- ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
- if (ret < 0)
- return ret;
- flt_data = c->flt_data;
- }
-
- if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
- /* convert input samples to fltp and scale to s16 range */
- ret = ff_audio_convert(c->ac_in, flt_data, src);
- if (ret < 0)
- return ret;
- } else if (c->apply_map) {
- ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
- if (ret < 0)
- return ret;
- } else {
- flt_data = src;
- }
-
- /* check alignment and padding constraints */
- if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align);
- int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
- int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align);
-
- if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
- c->quantize = c->ddsp.quantize;
- c->samples_align = c->ddsp.samples_align;
- } else {
- c->quantize = quantize_c;
- c->samples_align = 1;
- }
- }
-
- ret = convert_samples(c, (int16_t **)c->s16_data->data,
- (float * const *)flt_data->data, src->channels,
- src->nb_samples);
- if (ret < 0)
- return ret;
-
- c->s16_data->nb_samples = src->nb_samples;
-
- /* interleave output to dst if needed */
- if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
- ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
- if (ret < 0)
- return ret;
- } else
- c->s16_data = NULL;
-
- return 0;
-}
-
-void ff_dither_free(DitherContext **cp)
-{
- DitherContext *c = *cp;
- int ch;
-
- if (!c)
- return;
- ff_audio_data_free(&c->flt_data);
- ff_audio_data_free(&c->s16_data);
- ff_audio_convert_free(&c->ac_in);
- ff_audio_convert_free(&c->ac_out);
- for (ch = 0; ch < c->channels; ch++)
- av_free(c->state[ch].noise_buf);
- av_free(c->state);
- av_freep(cp);
-}
-
-static av_cold void dither_init(DitherDSPContext *ddsp,
- enum AVResampleDitherMethod method)
-{
- ddsp->quantize = quantize_c;
- ddsp->ptr_align = 1;
- ddsp->samples_align = 1;
-
- if (method == AV_RESAMPLE_DITHER_RECTANGULAR)
- ddsp->dither_int_to_float = dither_int_to_float_rectangular_c;
- else
- ddsp->dither_int_to_float = dither_int_to_float_triangular_c;
-
- if (ARCH_X86)
- ff_dither_init_x86(ddsp, method);
-}
-
-DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels, int sample_rate, int apply_map)
-{
- AVLFG seed_gen;
- DitherContext *c;
- int ch;
-
- if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
- av_get_bytes_per_sample(in_fmt) <= 2) {
- av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
- av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
- return NULL;
- }
-
- c = av_mallocz(sizeof(*c));
- if (!c)
- return NULL;
-
- c->apply_map = apply_map;
- if (apply_map)
- c->ch_map_info = &avr->ch_map_info;
-
- if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
- sample_rate != 48000 && sample_rate != 44100) {
- av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
- "for triangular_ns dither. using triangular_hp instead.\n");
- avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
- }
- c->method = avr->dither_method;
- dither_init(&c->ddsp, c->method);
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- if (sample_rate == 48000) {
- c->ns_coef_b = ns_48_coef_b;
- c->ns_coef_a = ns_48_coef_a;
- } else {
- c->ns_coef_b = ns_44_coef_b;
- c->ns_coef_a = ns_44_coef_a;
- }
- }
-
- /* Either s16 or s16p output format is allowed, but s16p is used
- internally, so we need to use a temp buffer and interleave if the output
- format is s16 */
- if (out_fmt != AV_SAMPLE_FMT_S16P) {
- c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
- "dither s16 buffer");
- if (!c->s16_data)
- goto fail;
-
- c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
- channels, sample_rate, 0);
- if (!c->ac_out)
- goto fail;
- }
-
- if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
- c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
- "dither flt buffer");
- if (!c->flt_data)
- goto fail;
- }
- if (in_fmt != AV_SAMPLE_FMT_FLTP) {
- c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
- channels, sample_rate, c->apply_map);
- if (!c->ac_in)
- goto fail;
- }
-
- c->state = av_mallocz(channels * sizeof(*c->state));
- if (!c->state)
- goto fail;
- c->channels = channels;
-
- /* calculate thresholds for turning off dithering during periods of
- silence to avoid replacing digital silence with quiet dither noise */
- c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
- c->mute_reset_threshold = c->mute_dither_threshold * 4;
-
- /* initialize dither states */
- av_lfg_init(&seed_gen, 0xC0FFEE);
- for (ch = 0; ch < channels; ch++) {
- DitherState *state = &c->state[ch];
- state->mute = c->mute_reset_threshold + 1;
- state->seed = av_lfg_get(&seed_gen);
- generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
- }
-
- return c;
-
-fail:
- ff_dither_free(&c);
- return NULL;
-}
diff --git a/libavresample/dither.h b/libavresample/dither.h
deleted file mode 100644
index 72f09cbdde..0000000000
--- a/libavresample/dither.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_DITHER_H
-#define AVRESAMPLE_DITHER_H
-
-#include "avresample.h"
-#include "audio_data.h"
-
-typedef struct DitherContext DitherContext;
-
-typedef struct DitherDSPContext {
- /**
- * Convert samples from flt to s16 with added dither noise.
- *
- * @param dst destination float array, range -0.5 to 0.5
- * @param src source int array, range INT_MIN to INT_MAX.
- * @param dither float dither noise array
- * @param len number of samples
- */
- void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
-
- int ptr_align; ///< src and dst constraints for quantize()
- int samples_align; ///< len constraints for quantize()
-
- /**
- * Convert dither noise from int to float with triangular distribution.
- *
- * @param dst destination float array, range -0.5 to 0.5
- * constraints: 32-byte aligned
- * @param src0 source int array, range INT_MIN to INT_MAX.
- * the array size is len * 2
- * constraints: 32-byte aligned
- * @param len number of output noise samples
- * constraints: multiple of 16
- */
- void (*dither_int_to_float)(float *dst, int *src0, int len);
-} DitherDSPContext;
-
-/**
- * Allocate and initialize a DitherContext.
- *
- * The parameters in the AVAudioResampleContext are used to initialize the
- * DitherContext.
- *
- * @param avr AVAudioResampleContext
- * @return newly-allocated DitherContext
- */
-DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels, int sample_rate, int apply_map);
-
-/**
- * Free a DitherContext.
- *
- * @param c DitherContext
- */
-void ff_dither_free(DitherContext **c);
-
-/**
- * Convert audio sample format with dithering.
- *
- * @param c DitherContext
- * @param dst destination audio data
- * @param src source audio data
- * @return 0 if ok, negative AVERROR code on failure
- */
-int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src);
-
-/* arch-specific initialization functions */
-
-void ff_dither_init_x86(DitherDSPContext *ddsp,
- enum AVResampleDitherMethod method);
-
-#endif /* AVRESAMPLE_DITHER_H */
diff --git a/libavresample/internal.h b/libavresample/internal.h
deleted file mode 100644
index 2fc3f6da67..0000000000
--- a/libavresample/internal.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 AVRESAMPLE_INTERNAL_H
-#define AVRESAMPLE_INTERNAL_H
-
-#include "libavutil/audio_fifo.h"
-#include "libavutil/log.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-#include "avresample.h"
-
-typedef struct AudioData AudioData;
-typedef struct AudioConvert AudioConvert;
-typedef struct AudioMix AudioMix;
-typedef struct ResampleContext ResampleContext;
-
-enum RemapPoint {
- REMAP_NONE,
- REMAP_IN_COPY,
- REMAP_IN_CONVERT,
- REMAP_OUT_COPY,
- REMAP_OUT_CONVERT,
-};
-
-typedef struct ChannelMapInfo {
- int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */
- int do_remap; /**< remap needed */
- int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */
- int do_copy; /**< copy needed */
- int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */
- int do_zero; /**< zeroing needed */
- int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */
-} ChannelMapInfo;
-
-struct AVAudioResampleContext {
- const AVClass *av_class; /**< AVClass for logging and AVOptions */
-
- uint64_t in_channel_layout; /**< input channel layout */
- enum AVSampleFormat in_sample_fmt; /**< input sample format */
- int in_sample_rate; /**< input sample rate */
- uint64_t out_channel_layout; /**< output channel layout */
- enum AVSampleFormat out_sample_fmt; /**< output sample format */
- int out_sample_rate; /**< output sample rate */
- enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
- enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
- double center_mix_level; /**< center mix level */
- double surround_mix_level; /**< surround mix level */
- double lfe_mix_level; /**< lfe mix level */
- int normalize_mix_level; /**< enable mix level normalization */
- int force_resampling; /**< force resampling */
- int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
- int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
- int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
- double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
- enum AVResampleFilterType filter_type; /**< resampling filter type */
- int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
- enum AVResampleDitherMethod dither_method; /**< dither method */
-
- int in_channels; /**< number of input channels */
- int out_channels; /**< number of output channels */
- int resample_channels; /**< number of channels used for resampling */
- int downmix_needed; /**< downmixing is needed */
- int upmix_needed; /**< upmixing is needed */
- int mixing_needed; /**< either upmixing or downmixing is needed */
- int resample_needed; /**< resampling is needed */
- int in_convert_needed; /**< input sample format conversion is needed */
- int out_convert_needed; /**< output sample format conversion is needed */
- int in_copy_needed; /**< input data copy is needed */
-
- AudioData *in_buffer; /**< buffer for converted input */
- AudioData *resample_out_buffer; /**< buffer for output from resampler */
- AudioData *out_buffer; /**< buffer for converted output */
- AVAudioFifo *out_fifo; /**< FIFO for output samples */
-
- AudioConvert *ac_in; /**< input sample format conversion context */
- AudioConvert *ac_out; /**< output sample format conversion context */
- ResampleContext *resample; /**< resampling context */
- AudioMix *am; /**< channel mixing context */
- enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */
-
- /**
- * mix matrix
- * only used if avresample_set_matrix() is called before avresample_open()
- */
- double *mix_matrix;
-
- int use_channel_map;
- enum RemapPoint remap_point;
- ChannelMapInfo ch_map_info;
-};
-
-
-void ff_audio_resample_init_aarch64(ResampleContext *c,
- enum AVSampleFormat sample_fmt);
-void ff_audio_resample_init_arm(ResampleContext *c,
- enum AVSampleFormat sample_fmt);
-
-#endif /* AVRESAMPLE_INTERNAL_H */
diff --git a/libavresample/libavresample.v b/libavresample/libavresample.v
deleted file mode 100644
index d6fc7512ba..0000000000
--- a/libavresample/libavresample.v
+++ /dev/null
@@ -1,6 +0,0 @@
-LIBAVRESAMPLE_MAJOR {
- global:
- av*;
- local:
- *;
-};
diff --git a/libavresample/options.c b/libavresample/options.c
deleted file mode 100644
index 5f08cd7e52..0000000000
--- a/libavresample/options.c
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-
-#include "libavutil/mathematics.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "avresample.h"
-#include "internal.h"
-#include "audio_mix.h"
-
-/**
- * @file
- * Options definition for AVAudioResampleContext.
- */
-
-#define OFFSET(x) offsetof(AVAudioResampleContext, x)
-#define PARAM AV_OPT_FLAG_AUDIO_PARAM
-
-static const AVOption avresample_options[] = {
- { "in_channel_layout", "Input Channel Layout", OFFSET(in_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM },
- { "in_sample_fmt", "Input Sample Format", OFFSET(in_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
- { "in_sample_rate", "Input Sample Rate", OFFSET(in_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM },
- { "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM },
- { "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
- { "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM },
- { "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM, "internal_sample_fmt" },
- {"u8" , "8-bit unsigned integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"s16", "16-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"s32", "32-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"flt", "32-bit float", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"dbl", "64-bit double", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"u8p" , "8-bit unsigned integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"s16p", "16-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"s32p", "32-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"fltp", "32-bit float planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLTP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- {"dblp", "64-bit double planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBLP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"},
- { "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { .i64 = AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
- { "q8", "16-bit 8.8 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q8 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
- { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
- { "flt", "Floating-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
- { "center_mix_level", "Center Mix Level", OFFSET(center_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM },
- { "surround_mix_level", "Surround Mix Level", OFFSET(surround_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM },
- { "lfe_mix_level", "LFE Mix Level", OFFSET(lfe_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -32.0, 32.0, PARAM },
- { "normalize_mix_level", "Normalize Mix Level", OFFSET(normalize_mix_level), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, PARAM },
- { "force_resampling", "Force Resampling", OFFSET(force_resampling), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM },
- { "filter_size", "Resampling Filter Size", OFFSET(filter_size), AV_OPT_TYPE_INT, { .i64 = 16 }, 0, 32, /* ??? */ PARAM },
- { "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { .i64 = 10 }, 0, 30, /* ??? */ PARAM },
- { "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM },
- { "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM },
- /* duplicate option in order to work with avconv */
- { "resample_cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM },
- { "matrix_encoding", "Matrixed Stereo Encoding", OFFSET(matrix_encoding), AV_OPT_TYPE_INT, {.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
- { "none", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
- { "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
- { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
- { "filter_type", "Filter Type", OFFSET(filter_type), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER, PARAM, "filter_type" },
- { "cubic", "Cubic", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
- { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
- { "kaiser", "Kaiser Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
- { "kaiser_beta", "Kaiser Window Beta", OFFSET(kaiser_beta), AV_OPT_TYPE_INT, { .i64 = 9 }, 2, 16, PARAM },
- { "dither_method", "Dither Method", OFFSET(dither_method), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_DITHER_NONE }, 0, AV_RESAMPLE_DITHER_NB-1, PARAM, "dither_method"},
- {"none", "No Dithering", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_NONE }, INT_MIN, INT_MAX, PARAM, "dither_method"},
- {"rectangular", "Rectangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_RECTANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"},
- {"triangular", "Triangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"},
- {"triangular_hp", "Triangular Dither With High Pass", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_HP }, INT_MIN, INT_MAX, PARAM, "dither_method"},
- {"triangular_ns", "Triangular Dither With Noise Shaping", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_NS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
- { NULL },
-};
-
-static const AVClass av_resample_context_class = {
- .class_name = "AVAudioResampleContext",
- .item_name = av_default_item_name,
- .option = avresample_options,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-AVAudioResampleContext *avresample_alloc_context(void)
-{
- AVAudioResampleContext *avr;
-
- avr = av_mallocz(sizeof(*avr));
- if (!avr)
- return NULL;
-
- avr->av_class = &av_resample_context_class;
- av_opt_set_defaults(avr);
-
- return avr;
-}
-
-const AVClass *avresample_get_class(void)
-{
- return &av_resample_context_class;
-}
diff --git a/libavresample/resample.c b/libavresample/resample.c
deleted file mode 100644
index dc14cc2d2a..0000000000
--- a/libavresample/resample.c
+++ /dev/null
@@ -1,446 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni at gmx.at>
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 "libavutil/common.h"
-#include "libavutil/libm.h"
-#include "libavutil/log.h"
-#include "internal.h"
-#include "resample.h"
-#include "audio_data.h"
-
-
-/* double template */
-#define CONFIG_RESAMPLE_DBL
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_DBL
-
-/* float template */
-#define CONFIG_RESAMPLE_FLT
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_FLT
-
-/* s32 template */
-#define CONFIG_RESAMPLE_S32
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_S32
-
-/* s16 template */
-#include "resample_template.c"
-
-
-/* 0th order modified Bessel function of the first kind. */
-static double bessel(double x)
-{
- double v = 1;
- double lastv = 0;
- double t = 1;
- int i;
-
- x = x * x / 4;
- for (i = 1; v != lastv; i++) {
- lastv = v;
- t *= x / (i * i);
- v += t;
- }
- return v;
-}
-
-/* Build a polyphase filterbank. */
-static int build_filter(ResampleContext *c, double factor)
-{
- int ph, i;
- double x, y, w;
- double *tab;
- int tap_count = c->filter_length;
- int phase_count = 1 << c->phase_shift;
- const int center = (tap_count - 1) / 2;
-
- tab = av_malloc(tap_count * sizeof(*tab));
- if (!tab)
- return AVERROR(ENOMEM);
-
- for (ph = 0; ph < phase_count; ph++) {
- double norm = 0;
- for (i = 0; i < tap_count; i++) {
- x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
- if (x == 0) y = 1.0;
- else y = sin(x) / x;
- switch (c->filter_type) {
- case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
- const float d = -0.5; //first order derivative = -0.5
- x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
- if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
- else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
- break;
- }
- case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
- w = 2.0 * x / (factor * tap_count) + M_PI;
- y *= 0.3635819 - 0.4891775 * cos( w) +
- 0.1365995 * cos(2 * w) -
- 0.0106411 * cos(3 * w);
- break;
- case AV_RESAMPLE_FILTER_TYPE_KAISER:
- w = 2.0 * x / (factor * tap_count * M_PI);
- y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
- break;
- }
-
- tab[i] = y;
- norm += y;
- }
- /* normalize so that an uniform color remains the same */
- for (i = 0; i < tap_count; i++)
- tab[i] = tab[i] / norm;
-
- c->set_filter(c->filter_bank, tab, ph, tap_count);
- }
-
- av_free(tab);
- return 0;
-}
-
-ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
-{
- ResampleContext *c;
- int out_rate = avr->out_sample_rate;
- int in_rate = avr->in_sample_rate;
- double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
- int phase_count = 1 << avr->phase_shift;
- int felem_size;
-
- if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
- av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
- "resampling: %s\n",
- av_get_sample_fmt_name(avr->internal_sample_fmt));
- return NULL;
- }
- c = av_mallocz(sizeof(*c));
- if (!c)
- return NULL;
-
- c->avr = avr;
- c->phase_shift = avr->phase_shift;
- c->phase_mask = phase_count - 1;
- c->linear = avr->linear_interp;
- c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
- c->filter_type = avr->filter_type;
- c->kaiser_beta = avr->kaiser_beta;
-
- switch (avr->internal_sample_fmt) {
- case AV_SAMPLE_FMT_DBLP:
- c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl;
- c->resample_nearest = resample_nearest_dbl;
- c->set_filter = set_filter_dbl;
- break;
- case AV_SAMPLE_FMT_FLTP:
- c->resample_one = c->linear ? resample_linear_flt : resample_one_flt;
- c->resample_nearest = resample_nearest_flt;
- c->set_filter = set_filter_flt;
- break;
- case AV_SAMPLE_FMT_S32P:
- c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32;
- c->resample_nearest = resample_nearest_s32;
- c->set_filter = set_filter_s32;
- break;
- case AV_SAMPLE_FMT_S16P:
- c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16;
- c->resample_nearest = resample_nearest_s16;
- c->set_filter = set_filter_s16;
- break;
- }
-
- if (ARCH_AARCH64)
- ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
- if (ARCH_ARM)
- ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
-
- felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
- c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
- if (!c->filter_bank)
- goto error;
-
- if (build_filter(c, factor) < 0)
- goto error;
-
- memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
- c->filter_bank, (c->filter_length - 1) * felem_size);
- memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
- &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
-
- c->compensation_distance = 0;
- if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
- in_rate * (int64_t)phase_count, INT32_MAX / 2))
- goto error;
- c->ideal_dst_incr = c->dst_incr;
-
- c->padding_size = (c->filter_length - 1) / 2;
- c->initial_padding_filled = 0;
- c->index = 0;
- c->frac = 0;
-
- /* allocate internal buffer */
- c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
- avr->internal_sample_fmt,
- "resample buffer");
- if (!c->buffer)
- goto error;
- c->buffer->nb_samples = c->padding_size;
- c->initial_padding_samples = c->padding_size;
-
- av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
- av_get_sample_fmt_name(avr->internal_sample_fmt),
- avr->in_sample_rate, avr->out_sample_rate);
-
- return c;
-
-error:
- ff_audio_data_free(&c->buffer);
- av_free(c->filter_bank);
- av_free(c);
- return NULL;
-}
-
-void ff_audio_resample_free(ResampleContext **c)
-{
- if (!*c)
- return;
- ff_audio_data_free(&(*c)->buffer);
- av_free((*c)->filter_bank);
- av_freep(c);
-}
-
-int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
- int compensation_distance)
-{
- ResampleContext *c;
-
- if (compensation_distance < 0)
- return AVERROR(EINVAL);
- if (!compensation_distance && sample_delta)
- return AVERROR(EINVAL);
-
- if (!avr->resample_needed) {
- av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
- return AVERROR(EINVAL);
- }
- c = avr->resample;
- c->compensation_distance = compensation_distance;
- if (compensation_distance) {
- c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
- (int64_t)sample_delta / compensation_distance;
- } else {
- c->dst_incr = c->ideal_dst_incr;
- }
-
- return 0;
-}
-
-static int resample(ResampleContext *c, void *dst, const void *src,
- int *consumed, int src_size, int dst_size, int update_ctx,
- int nearest_neighbour)
-{
- int dst_index;
- unsigned int index = c->index;
- int frac = c->frac;
- int dst_incr_frac = c->dst_incr % c->src_incr;
- int dst_incr = c->dst_incr / c->src_incr;
- int compensation_distance = c->compensation_distance;
-
- if (!dst != !src)
- return AVERROR(EINVAL);
-
- if (nearest_neighbour) {
- uint64_t index2 = ((uint64_t)index) << 32;
- int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
- dst_size = FFMIN(dst_size,
- (src_size-1-index) * (int64_t)c->src_incr /
- c->dst_incr);
-
- if (dst) {
- for(dst_index = 0; dst_index < dst_size; dst_index++) {
- c->resample_nearest(dst, dst_index, src, index2 >> 32);
- index2 += incr;
- }
- } else {
- dst_index = dst_size;
- }
- index += dst_index * dst_incr;
- index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
- frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
- } else {
- for (dst_index = 0; dst_index < dst_size; dst_index++) {
- int sample_index = index >> c->phase_shift;
-
- if (sample_index + c->filter_length > src_size)
- break;
-
- if (dst)
- c->resample_one(c, dst, dst_index, src, index, frac);
-
- frac += dst_incr_frac;
- index += dst_incr;
- if (frac >= c->src_incr) {
- frac -= c->src_incr;
- index++;
- }
- if (dst_index + 1 == compensation_distance) {
- compensation_distance = 0;
- dst_incr_frac = c->ideal_dst_incr % c->src_incr;
- dst_incr = c->ideal_dst_incr / c->src_incr;
- }
- }
- }
- if (consumed)
- *consumed = index >> c->phase_shift;
-
- if (update_ctx) {
- index &= c->phase_mask;
-
- if (compensation_distance) {
- compensation_distance -= dst_index;
- if (compensation_distance <= 0)
- return AVERROR_BUG;
- }
- c->frac = frac;
- c->index = index;
- c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
- c->compensation_distance = compensation_distance;
- }
-
- return dst_index;
-}
-
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
-{
- int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
- int ret = AVERROR(EINVAL);
- int nearest_neighbour = (c->compensation_distance == 0 &&
- c->filter_length == 1 &&
- c->phase_shift == 0);
-
- in_samples = src ? src->nb_samples : 0;
- in_leftover = c->buffer->nb_samples;
-
- /* add input samples to the internal buffer */
- if (src) {
- ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
- if (ret < 0)
- return ret;
- } else if (in_leftover <= c->final_padding_samples) {
- /* no remaining samples to flush */
- return 0;
- }
-
- if (!c->initial_padding_filled) {
- int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
- int i;
-
- if (src && c->buffer->nb_samples < 2 * c->padding_size)
- return 0;
-
- for (i = 0; i < c->padding_size; i++)
- for (ch = 0; ch < c->buffer->channels; ch++) {
- if (c->buffer->nb_samples > 2 * c->padding_size - i) {
- memcpy(c->buffer->data[ch] + bps * i,
- c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
- } else {
- memset(c->buffer->data[ch] + bps * i, 0, bps);
- }
- }
- c->initial_padding_filled = 1;
- }
-
- if (!src && !c->final_padding_filled) {
- int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
- int i;
-
- ret = ff_audio_data_realloc(c->buffer,
- FFMAX(in_samples, in_leftover) +
- c->padding_size);
- if (ret < 0) {
- av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
- return AVERROR(ENOMEM);
- }
-
- for (i = 0; i < c->padding_size; i++)
- for (ch = 0; ch < c->buffer->channels; ch++) {
- if (in_leftover > i) {
- memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
- c->buffer->data[ch] + bps * (in_leftover - i - 1),
- bps);
- } else {
- memset(c->buffer->data[ch] + bps * (in_leftover + i),
- 0, bps);
- }
- }
- c->buffer->nb_samples += c->padding_size;
- c->final_padding_samples = c->padding_size;
- c->final_padding_filled = 1;
- }
-
-
- /* calculate output size and reallocate output buffer if needed */
- /* TODO: try to calculate this without the dummy resample() run */
- if (!dst->read_only && dst->allow_realloc) {
- out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
- INT_MAX, 0, nearest_neighbour);
- ret = ff_audio_data_realloc(dst, out_samples);
- if (ret < 0) {
- av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
- return ret;
- }
- }
-
- /* resample each channel plane */
- for (ch = 0; ch < c->buffer->channels; ch++) {
- out_samples = resample(c, (void *)dst->data[ch],
- (const void *)c->buffer->data[ch], &consumed,
- c->buffer->nb_samples, dst->allocated_samples,
- ch + 1 == c->buffer->channels, nearest_neighbour);
- }
- if (out_samples < 0) {
- av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
- return out_samples;
- }
-
- /* drain consumed samples from the internal buffer */
- ff_audio_data_drain(c->buffer, consumed);
- c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
-
- av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
- in_samples, in_leftover, out_samples, c->buffer->nb_samples);
-
- dst->nb_samples = out_samples;
- return 0;
-}
-
-int avresample_get_delay(AVAudioResampleContext *avr)
-{
- ResampleContext *c = avr->resample;
-
- if (!avr->resample_needed || !avr->resample)
- return 0;
-
- return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
-}
diff --git a/libavresample/resample.h b/libavresample/resample.h
deleted file mode 100644
index be9f562791..0000000000
--- a/libavresample/resample.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni at gmx.at>
- *
- * 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 AVRESAMPLE_RESAMPLE_H
-#define AVRESAMPLE_RESAMPLE_H
-
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-
-struct ResampleContext {
- AVAudioResampleContext *avr;
- AudioData *buffer;
- uint8_t *filter_bank;
- int filter_length;
- int ideal_dst_incr;
- int dst_incr;
- unsigned int index;
- int frac;
- int src_incr;
- int compensation_distance;
- int phase_shift;
- int phase_mask;
- int linear;
- enum AVResampleFilterType filter_type;
- int kaiser_beta;
- void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
- void (*resample_one)(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0,
- unsigned int index, int frac);
- void (*resample_nearest)(void *dst0, int dst_index,
- const void *src0, unsigned int index);
- int padding_size;
- int initial_padding_filled;
- int initial_padding_samples;
- int final_padding_filled;
- int final_padding_samples;
-};
-
-/**
- * Allocate and initialize a ResampleContext.
- *
- * The parameters in the AVAudioResampleContext are used to initialize the
- * ResampleContext.
- *
- * @param avr AVAudioResampleContext
- * @return newly-allocated ResampleContext
- */
-ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
-
-/**
- * Free a ResampleContext.
- *
- * @param c ResampleContext
- */
-void ff_audio_resample_free(ResampleContext **c);
-
-/**
- * Resample audio data.
- *
- * Changes the sample rate.
- *
- * @par
- * All samples in the source data may not be consumed depending on the
- * resampling parameters and the size of the output buffer. The unconsumed
- * samples are automatically added to the start of the source in the next call.
- * If the destination data can be reallocated, that may be done in this function
- * in order to fit all available output. If it cannot be reallocated, fewer
- * input samples will be consumed in order to have the output fit in the
- * destination data buffers.
- *
- * @param c ResampleContext
- * @param dst destination audio data
- * @param src source audio data
- * @return 0 on success, negative AVERROR code on failure
- */
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src);
-
-#endif /* AVRESAMPLE_RESAMPLE_H */
diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c
deleted file mode 100644
index 863852a3fd..0000000000
--- a/libavresample/resample_template.c
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni at gmx.at>
- *
- * 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 <math.h>
-#include <stdint.h>
-
-#include "libavutil/common.h"
-#include "internal.h"
-
-#if defined(CONFIG_RESAMPLE_DBL)
-#define SET_TYPE(func) func ## _dbl
-#define FELEM double
-#define FELEM2 double
-#define FELEML double
-#define OUT(d, v) d = v
-#define DBL_TO_FELEM(d, v) d = v
-#elif defined(CONFIG_RESAMPLE_FLT)
-#define SET_TYPE(func) func ## _flt
-#define FELEM float
-#define FELEM2 float
-#define FELEML float
-#define OUT(d, v) d = v
-#define DBL_TO_FELEM(d, v) d = v
-#elif defined(CONFIG_RESAMPLE_S32)
-#define SET_TYPE(func) func ## _s32
-#define FELEM int32_t
-#define FELEM2 int64_t
-#define FELEML int64_t
-#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
-#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
-#else
-#define SET_TYPE(func) func ## _s16
-#define FELEM int16_t
-#define FELEM2 int32_t
-#define FELEML int64_t
-#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
-#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
-#endif
-
-static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index)
-{
- FELEM *dst = dst0;
- const FELEM *src = src0;
- dst[dst_index] = src[index];
-}
-
-static void SET_TYPE(resample_linear)(ResampleContext *c, void *dst0, int dst_index,
- const void *src0, unsigned int index, int frac)
-{
- FELEM *dst = dst0;
- const FELEM *src = src0;
- int i;
- unsigned int sample_index = index >> c->phase_shift;
- FELEM2 val = 0;
- FELEM *filter = ((FELEM *)c->filter_bank) +
- c->filter_length * (index & c->phase_mask);
- FELEM2 v2 = 0;
-
- for (i = 0; i < c->filter_length; i++) {
- val += src[sample_index + i] * (FELEM2)filter[i];
- v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
- }
- val += (v2 - val) * (FELEML)frac / c->src_incr;
-
- OUT(dst[dst_index], val);
-}
-
-static void SET_TYPE(resample_one)(ResampleContext *c,
- void *dst0, int dst_index, const void *src0,
- unsigned int index, int frac)
-{
- FELEM *dst = dst0;
- const FELEM *src = src0;
- int i;
- unsigned int sample_index = index >> c->phase_shift;
- FELEM2 val = 0;
- FELEM *filter = ((FELEM *)c->filter_bank) +
- c->filter_length * (index & c->phase_mask);
-
- for (i = 0; i < c->filter_length; i++)
- val += src[sample_index + i] * (FELEM2)filter[i];
-
- OUT(dst[dst_index], val);
-}
-
-static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
- int tap_count)
-{
- int i;
- FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
- for (i = 0; i < tap_count; i++) {
- DBL_TO_FELEM(filter[i], tab[i]);
- }
-}
-
-#undef SET_TYPE
-#undef FELEM
-#undef FELEM2
-#undef FELEML
-#undef OUT
-#undef DBL_TO_FELEM
diff --git a/libavresample/tests/.gitignore b/libavresample/tests/.gitignore
deleted file mode 100644
index 1e15871d54..0000000000
--- a/libavresample/tests/.gitignore
+++ /dev/null
@@ -1 +0,0 @@
-/avresample
diff --git a/libavresample/tests/avresample.c b/libavresample/tests/avresample.c
deleted file mode 100644
index 8c377bae84..0000000000
--- a/libavresample/tests/avresample.c
+++ /dev/null
@@ -1,342 +0,0 @@
-/*
- * Copyright (c) 2002 Fabrice Bellard
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 <stdint.h>
-#include <stdio.h>
-
-#include "libavutil/avstring.h"
-#include "libavutil/common.h"
-#include "libavutil/lfg.h"
-#include "libavutil/libm.h"
-#include "libavutil/log.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavresample/avresample.h"
-
-static double dbl_rand(AVLFG *lfg)
-{
- return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
-}
-
-#define PUT_FUNC(name, fmt, type, expr) \
-static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
- int channels, int sample, int ch, \
- double v_dbl) \
-{ \
- type v = expr; \
- type **out = (type **)data; \
- if (av_sample_fmt_is_planar(sample_fmt)) \
- out[ch][sample] = v; \
- else \
- out[0][sample * channels + ch] = v; \
-}
-
-PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
-PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
-PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
-PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
-PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
-
-static void put_sample(void **data, enum AVSampleFormat sample_fmt,
- int channels, int sample, int ch, double v_dbl)
-{
- switch (av_get_packed_sample_fmt(sample_fmt)) {
- case AV_SAMPLE_FMT_U8:
- put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
- break;
- case AV_SAMPLE_FMT_S16:
- put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
- break;
- case AV_SAMPLE_FMT_S32:
- put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
- break;
- case AV_SAMPLE_FMT_FLT:
- put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
- break;
- case AV_SAMPLE_FMT_DBL:
- put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
- break;
- }
-}
-
-static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
- int channels, int sample_rate, int nb_samples)
-{
- int i, ch, k;
- double v, f, a, ampa;
- double tabf1[AVRESAMPLE_MAX_CHANNELS];
- double tabf2[AVRESAMPLE_MAX_CHANNELS];
- double taba[AVRESAMPLE_MAX_CHANNELS];
-
-#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
-
- k = 0;
-
- /* 1 second of single freq sine at 1000 Hz */
- a = 0;
- for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
- v = sin(a) * 0.30;
- for (ch = 0; ch < channels; ch++)
- PUT_SAMPLE
- a += M_PI * 1000.0 * 2.0 / sample_rate;
- }
-
- /* 1 second of varying frequency between 100 and 10000 Hz */
- a = 0;
- for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
- v = sin(a) * 0.30;
- for (ch = 0; ch < channels; ch++)
- PUT_SAMPLE
- f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
- a += M_PI * f * 2.0 / sample_rate;
- }
-
- /* 0.5 second of low amplitude white noise */
- for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
- v = dbl_rand(rnd) * 0.30;
- for (ch = 0; ch < channels; ch++)
- PUT_SAMPLE
- }
-
- /* 0.5 second of high amplitude white noise */
- for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
- v = dbl_rand(rnd);
- for (ch = 0; ch < channels; ch++)
- PUT_SAMPLE
- }
-
- /* 1 second of unrelated ramps for each channel */
- for (ch = 0; ch < channels; ch++) {
- taba[ch] = 0;
- tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
- tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
- }
- for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
- for (ch = 0; ch < channels; ch++) {
- v = sin(taba[ch]) * 0.30;
- PUT_SAMPLE
- f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
- taba[ch] += M_PI * f * 2.0 / sample_rate;
- }
- }
-
- /* 2 seconds of 500 Hz with varying volume */
- a = 0;
- ampa = 0;
- for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
- for (ch = 0; ch < channels; ch++) {
- double amp = (1.0 + sin(ampa)) * 0.15;
- if (ch & 1)
- amp = 0.30 - amp;
- v = sin(a) * amp;
- PUT_SAMPLE
- a += M_PI * 500.0 * 2.0 / sample_rate;
- ampa += M_PI * 2.0 / sample_rate;
- }
- }
-}
-
-/* formats, rates, and layouts are ordered for priority in testing.
- e.g. 'avresample-test 4 2 2' will test all input/output combinations of
- S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
-
-static const enum AVSampleFormat formats[] = {
- AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_S16P,
- AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S32P,
- AV_SAMPLE_FMT_S32,
- AV_SAMPLE_FMT_U8P,
- AV_SAMPLE_FMT_U8,
- AV_SAMPLE_FMT_DBLP,
- AV_SAMPLE_FMT_DBL,
-};
-
-static const int rates[] = {
- 48000,
- 44100,
- 16000
-};
-
-static const uint64_t layouts[] = {
- AV_CH_LAYOUT_STEREO,
- AV_CH_LAYOUT_MONO,
- AV_CH_LAYOUT_5POINT1,
- AV_CH_LAYOUT_7POINT1,
-};
-
-int main(int argc, char **argv)
-{
- AVAudioResampleContext *s;
- AVLFG rnd;
- int ret = 0;
- uint8_t *in_buf = NULL;
- uint8_t *out_buf = NULL;
- unsigned int in_buf_size;
- unsigned int out_buf_size;
- uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
- uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
- int in_linesize;
- int out_linesize;
- uint64_t in_ch_layout;
- int in_channels;
- enum AVSampleFormat in_fmt;
- int in_rate;
- uint64_t out_ch_layout;
- int out_channels;
- enum AVSampleFormat out_fmt;
- int out_rate;
- int num_formats, num_rates, num_layouts;
- int i, j, k, l, m, n;
-
- num_formats = 2;
- num_rates = 2;
- num_layouts = 2;
- if (argc > 1) {
- if (!av_strncasecmp(argv[1], "-h", 3)) {
- av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
- "[<num sample rates> [<num channel layouts>]]]\n"
- "Default is 2 2 2\n");
- return 0;
- }
- num_formats = strtol(argv[1], NULL, 0);
- num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
- }
- if (argc > 2) {
- num_rates = strtol(argv[2], NULL, 0);
- num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
- }
- if (argc > 3) {
- num_layouts = strtol(argv[3], NULL, 0);
- num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
- }
-
- av_log_set_level(AV_LOG_DEBUG);
-
- av_lfg_init(&rnd, 0xC0FFEE);
-
- in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
- AV_SAMPLE_FMT_DBLP, 0);
- out_buf_size = in_buf_size;
-
- in_buf = av_malloc(in_buf_size);
- if (!in_buf)
- goto end;
- out_buf = av_malloc(out_buf_size);
- if (!out_buf)
- goto end;
-
- s = avresample_alloc_context();
- if (!s) {
- av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
- ret = 1;
- goto end;
- }
-
- for (i = 0; i < num_formats; i++) {
- in_fmt = formats[i];
- for (k = 0; k < num_layouts; k++) {
- in_ch_layout = layouts[k];
- in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
- for (m = 0; m < num_rates; m++) {
- in_rate = rates[m];
-
- ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
- in_channels, in_rate * 6,
- in_fmt, 0);
- if (ret < 0) {
- av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
- goto end;
- }
- audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
-
- for (j = 0; j < num_formats; j++) {
- out_fmt = formats[j];
- for (l = 0; l < num_layouts; l++) {
- out_ch_layout = layouts[l];
- out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
- for (n = 0; n < num_rates; n++) {
- out_rate = rates[n];
-
- av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
- av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
- in_channels, out_channels, in_rate, out_rate);
-
- ret = av_samples_fill_arrays(out_data, &out_linesize,
- out_buf, out_channels,
- out_rate * 6, out_fmt, 0);
- if (ret < 0) {
- av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
- goto end;
- }
-
- av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
- av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
- av_opt_set_int(s, "in_sample_rate", in_rate, 0);
- av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
- av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
- av_opt_set_int(s, "out_sample_rate", out_rate, 0);
-
- av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
-
- ret = avresample_open(s);
- if (ret < 0) {
- av_log(s, AV_LOG_ERROR, "Error opening context\n");
- goto end;
- }
-
- ret = avresample_convert(s, out_data, out_linesize, out_rate * 6,
- in_data, in_linesize, in_rate * 6);
- if (ret < 0) {
- char errbuf[256];
- av_strerror(ret, errbuf, sizeof(errbuf));
- av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
- goto end;
- }
- av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
- in_rate * 6, ret);
- if (avresample_get_delay(s) > 0)
- av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
- avresample_get_delay(s));
- if (avresample_available(s) > 0)
- av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
- avresample_available(s));
- av_log(NULL, AV_LOG_INFO, "\n");
-
- avresample_close(s);
- }
- }
- }
- }
- }
- }
-
- ret = 0;
-
-end:
- av_freep(&in_buf);
- av_freep(&out_buf);
- avresample_free(&s);
- return ret;
-}
diff --git a/libavresample/utils.c b/libavresample/utils.c
deleted file mode 100644
index b4fb906556..0000000000
--- a/libavresample/utils.c
+++ /dev/null
@@ -1,793 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 "libavutil/common.h"
-#include "libavutil/dict.h"
-// #include "libavutil/error.h"
-#include "libavutil/frame.h"
-#include "libavutil/log.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-
-#include "avresample.h"
-#include "internal.h"
-#include "audio_data.h"
-#include "audio_convert.h"
-#include "audio_mix.h"
-#include "resample.h"
-
-int avresample_open(AVAudioResampleContext *avr)
-{
- int ret;
-
- if (avresample_is_open(avr)) {
- av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
- return AVERROR(EINVAL);
- }
-
- /* set channel mixing parameters */
- avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
- if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
- avr->in_channel_layout);
- return AVERROR(EINVAL);
- }
- avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
- if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
- avr->out_channel_layout);
- return AVERROR(EINVAL);
- }
- avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
- avr->downmix_needed = avr->in_channels > avr->out_channels;
- avr->upmix_needed = avr->out_channels > avr->in_channels ||
- (!avr->downmix_needed && (avr->mix_matrix ||
- avr->in_channel_layout != avr->out_channel_layout));
- avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
-
- /* set resampling parameters */
- avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
- avr->force_resampling;
-
- /* select internal sample format if not specified by the user */
- if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
- (avr->mixing_needed || avr->resample_needed)) {
- enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
- enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
- int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
- av_get_bytes_per_sample(out_fmt));
- if (max_bps <= 2) {
- avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
- } else if (avr->mixing_needed) {
- avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
- } else {
- if (max_bps <= 4) {
- if (in_fmt == AV_SAMPLE_FMT_S32P ||
- out_fmt == AV_SAMPLE_FMT_S32P) {
- if (in_fmt == AV_SAMPLE_FMT_FLTP ||
- out_fmt == AV_SAMPLE_FMT_FLTP) {
- /* if one is s32 and the other is flt, use dbl */
- avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
- } else {
- /* if one is s32 and the other is s32, s16, or u8, use s32 */
- avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
- }
- } else {
- /* if one is flt and the other is flt, s16 or u8, use flt */
- avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
- }
- } else {
- /* if either is dbl, use dbl */
- avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
- }
- }
- av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
- av_get_sample_fmt_name(avr->internal_sample_fmt));
- }
-
- /* we may need to add an extra conversion in order to remap channels if
- the output format is not planar */
- if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
- !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels)) {
- avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
- }
-
- /* set sample format conversion parameters */
- if (avr->resample_needed || avr->mixing_needed)
- avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
- else
- avr->in_convert_needed = avr->use_channel_map &&
- !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels);
-
- if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
- avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
- else
- avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
-
- avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
- (avr->use_channel_map && avr->resample_needed));
-
- if (avr->use_channel_map) {
- if (avr->in_copy_needed) {
- avr->remap_point = REMAP_IN_COPY;
- av_log(avr, AV_LOG_TRACE, "remap channels during in_copy\n");
- } else if (avr->in_convert_needed) {
- avr->remap_point = REMAP_IN_CONVERT;
- av_log(avr, AV_LOG_TRACE, "remap channels during in_convert\n");
- } else if (avr->out_convert_needed) {
- avr->remap_point = REMAP_OUT_CONVERT;
- av_log(avr, AV_LOG_TRACE, "remap channels during out_convert\n");
- } else {
- avr->remap_point = REMAP_OUT_COPY;
- av_log(avr, AV_LOG_TRACE, "remap channels during out_copy\n");
- }
-
-#ifdef DEBUG
- {
- int ch;
- av_log(avr, AV_LOG_TRACE, "output map: ");
- if (avr->ch_map_info.do_remap)
- for (ch = 0; ch < avr->in_channels; ch++)
- av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_map[ch]);
- else
- av_log(avr, AV_LOG_TRACE, "n/a");
- av_log(avr, AV_LOG_TRACE, "\n");
- av_log(avr, AV_LOG_TRACE, "copy map: ");
- if (avr->ch_map_info.do_copy)
- for (ch = 0; ch < avr->in_channels; ch++)
- av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_copy[ch]);
- else
- av_log(avr, AV_LOG_TRACE, "n/a");
- av_log(avr, AV_LOG_TRACE, "\n");
- av_log(avr, AV_LOG_TRACE, "zero map: ");
- if (avr->ch_map_info.do_zero)
- for (ch = 0; ch < avr->in_channels; ch++)
- av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_zero[ch]);
- else
- av_log(avr, AV_LOG_TRACE, "n/a");
- av_log(avr, AV_LOG_TRACE, "\n");
- av_log(avr, AV_LOG_TRACE, "input map: ");
- for (ch = 0; ch < avr->in_channels; ch++)
- av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.input_map[ch]);
- av_log(avr, AV_LOG_TRACE, "\n");
- }
-#endif
- } else
- avr->remap_point = REMAP_NONE;
-
- /* allocate buffers */
- if (avr->in_copy_needed || avr->in_convert_needed) {
- avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
- 0, avr->internal_sample_fmt,
- "in_buffer");
- if (!avr->in_buffer) {
- ret = AVERROR(EINVAL);
- goto error;
- }
- }
- if (avr->resample_needed) {
- avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
- 1024, avr->internal_sample_fmt,
- "resample_out_buffer");
- if (!avr->resample_out_buffer) {
- ret = AVERROR(EINVAL);
- goto error;
- }
- }
- if (avr->out_convert_needed) {
- avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
- avr->out_sample_fmt, "out_buffer");
- if (!avr->out_buffer) {
- ret = AVERROR(EINVAL);
- goto error;
- }
- }
- avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
- 1024);
- if (!avr->out_fifo) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
-
- /* setup contexts */
- if (avr->in_convert_needed) {
- avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
- avr->in_sample_fmt, avr->in_channels,
- avr->in_sample_rate,
- avr->remap_point == REMAP_IN_CONVERT);
- if (!avr->ac_in) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
- }
- if (avr->out_convert_needed) {
- enum AVSampleFormat src_fmt;
- if (avr->in_convert_needed)
- src_fmt = avr->internal_sample_fmt;
- else
- src_fmt = avr->in_sample_fmt;
- avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
- avr->out_channels,
- avr->out_sample_rate,
- avr->remap_point == REMAP_OUT_CONVERT);
- if (!avr->ac_out) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
- }
- if (avr->resample_needed) {
- avr->resample = ff_audio_resample_init(avr);
- if (!avr->resample) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
- }
- if (avr->mixing_needed) {
- avr->am = ff_audio_mix_alloc(avr);
- if (!avr->am) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
- }
-
- return 0;
-
-error:
- avresample_close(avr);
- return ret;
-}
-
-int avresample_is_open(AVAudioResampleContext *avr)
-{
- return !!avr->out_fifo;
-}
-
-void avresample_close(AVAudioResampleContext *avr)
-{
- ff_audio_data_free(&avr->in_buffer);
- ff_audio_data_free(&avr->resample_out_buffer);
- ff_audio_data_free(&avr->out_buffer);
- av_audio_fifo_free(avr->out_fifo);
- avr->out_fifo = NULL;
- ff_audio_convert_free(&avr->ac_in);
- ff_audio_convert_free(&avr->ac_out);
- ff_audio_resample_free(&avr->resample);
- ff_audio_mix_free(&avr->am);
- av_freep(&avr->mix_matrix);
-
- avr->use_channel_map = 0;
-}
-
-void avresample_free(AVAudioResampleContext **avr)
-{
- if (!*avr)
- return;
- avresample_close(*avr);
- av_opt_free(*avr);
- av_freep(avr);
-}
-
-static int handle_buffered_output(AVAudioResampleContext *avr,
- AudioData *output, AudioData *converted)
-{
- int ret;
-
- if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
- (converted && output->allocated_samples < converted->nb_samples)) {
- if (converted) {
- /* if there are any samples in the output FIFO or if the
- user-supplied output buffer is not large enough for all samples,
- we add to the output FIFO */
- av_log(avr, AV_LOG_TRACE, "[FIFO] add %s to out_fifo\n", converted->name);
- ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
- converted->nb_samples);
- if (ret < 0)
- return ret;
- }
-
- /* if the user specified an output buffer, read samples from the output
- FIFO to the user output */
- if (output && output->allocated_samples > 0) {
- av_log(avr, AV_LOG_TRACE, "[FIFO] read from out_fifo to output\n");
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return ff_audio_data_read_from_fifo(avr->out_fifo, output,
- output->allocated_samples);
- }
- } else if (converted) {
- /* copy directly to output if it is large enough or there is not any
- data in the output FIFO */
- av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", converted->name);
- output->nb_samples = 0;
- ret = ff_audio_data_copy(output, converted,
- avr->remap_point == REMAP_OUT_COPY ?
- &avr->ch_map_info : NULL);
- if (ret < 0)
- return ret;
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return output->nb_samples;
- }
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return 0;
-}
-
-int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
- uint8_t **output, int out_plane_size,
- int out_samples,
- uint8_t * const *input,
- int in_plane_size, int in_samples)
-{
- AudioData input_buffer;
- AudioData output_buffer;
- AudioData *current_buffer;
- int ret, direct_output;
-
- /* reset internal buffers */
- if (avr->in_buffer) {
- avr->in_buffer->nb_samples = 0;
- ff_audio_data_set_channels(avr->in_buffer,
- avr->in_buffer->allocated_channels);
- }
- if (avr->resample_out_buffer) {
- avr->resample_out_buffer->nb_samples = 0;
- ff_audio_data_set_channels(avr->resample_out_buffer,
- avr->resample_out_buffer->allocated_channels);
- }
- if (avr->out_buffer) {
- avr->out_buffer->nb_samples = 0;
- ff_audio_data_set_channels(avr->out_buffer,
- avr->out_buffer->allocated_channels);
- }
-
- av_log(avr, AV_LOG_TRACE, "[start conversion]\n");
-
- /* initialize output_buffer with output data */
- direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
- if (output) {
- ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
- avr->out_channels, out_samples,
- avr->out_sample_fmt, 0, "output");
- if (ret < 0)
- return ret;
- output_buffer.nb_samples = 0;
- }
-
- if (input) {
- /* initialize input_buffer with input data */
- ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
- avr->in_channels, in_samples,
- avr->in_sample_fmt, 1, "input");
- if (ret < 0)
- return ret;
- current_buffer = &input_buffer;
-
- if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
- !avr->out_convert_needed && direct_output && out_samples >= in_samples) {
- /* in some rare cases we can copy input to output and upmix
- directly in the output buffer */
- av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", current_buffer->name);
- ret = ff_audio_data_copy(&output_buffer, current_buffer,
- avr->remap_point == REMAP_OUT_COPY ?
- &avr->ch_map_info : NULL);
- if (ret < 0)
- return ret;
- current_buffer = &output_buffer;
- } else if (avr->remap_point == REMAP_OUT_COPY &&
- (!direct_output || out_samples < in_samples)) {
- /* if remapping channels during output copy, we may need to
- * use an intermediate buffer in order to remap before adding
- * samples to the output fifo */
- av_log(avr, AV_LOG_TRACE, "[copy] %s to out_buffer\n", current_buffer->name);
- ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
- &avr->ch_map_info);
- if (ret < 0)
- return ret;
- current_buffer = avr->out_buffer;
- } else if (avr->in_copy_needed || avr->in_convert_needed) {
- /* if needed, copy or convert input to in_buffer, and downmix if
- applicable */
- if (avr->in_convert_needed) {
- ret = ff_audio_data_realloc(avr->in_buffer,
- current_buffer->nb_samples);
- if (ret < 0)
- return ret;
- av_log(avr, AV_LOG_TRACE, "[convert] %s to in_buffer\n", current_buffer->name);
- ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
- current_buffer);
- if (ret < 0)
- return ret;
- } else {
- av_log(avr, AV_LOG_TRACE, "[copy] %s to in_buffer\n", current_buffer->name);
- ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
- avr->remap_point == REMAP_IN_COPY ?
- &avr->ch_map_info : NULL);
- if (ret < 0)
- return ret;
- }
- ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
- if (avr->downmix_needed) {
- av_log(avr, AV_LOG_TRACE, "[downmix] in_buffer\n");
- ret = ff_audio_mix(avr->am, avr->in_buffer);
- if (ret < 0)
- return ret;
- }
- current_buffer = avr->in_buffer;
- }
- } else {
- /* flush resampling buffer and/or output FIFO if input is NULL */
- if (!avr->resample_needed)
- return handle_buffered_output(avr, output ? &output_buffer : NULL,
- NULL);
- current_buffer = NULL;
- }
-
- if (avr->resample_needed) {
- AudioData *resample_out;
-
- if (!avr->out_convert_needed && direct_output && out_samples > 0)
- resample_out = &output_buffer;
- else
- resample_out = avr->resample_out_buffer;
- av_log(avr, AV_LOG_TRACE, "[resample] %s to %s\n",
- current_buffer ? current_buffer->name : "null",
- resample_out->name);
- ret = ff_audio_resample(avr->resample, resample_out,
- current_buffer);
- if (ret < 0)
- return ret;
-
- /* if resampling did not produce any samples, just return 0 */
- if (resample_out->nb_samples == 0) {
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return 0;
- }
-
- current_buffer = resample_out;
- }
-
- if (avr->upmix_needed) {
- av_log(avr, AV_LOG_TRACE, "[upmix] %s\n", current_buffer->name);
- ret = ff_audio_mix(avr->am, current_buffer);
- if (ret < 0)
- return ret;
- }
-
- /* if we resampled or upmixed directly to output, return here */
- if (current_buffer == &output_buffer) {
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return current_buffer->nb_samples;
- }
-
- if (avr->out_convert_needed) {
- if (direct_output && out_samples >= current_buffer->nb_samples) {
- /* convert directly to output */
- av_log(avr, AV_LOG_TRACE, "[convert] %s to output\n", current_buffer->name);
- ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
- if (ret < 0)
- return ret;
-
- av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
- return output_buffer.nb_samples;
- } else {
- ret = ff_audio_data_realloc(avr->out_buffer,
- current_buffer->nb_samples);
- if (ret < 0)
- return ret;
- av_log(avr, AV_LOG_TRACE, "[convert] %s to out_buffer\n", current_buffer->name);
- ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
- current_buffer);
- if (ret < 0)
- return ret;
- current_buffer = avr->out_buffer;
- }
- }
-
- return handle_buffered_output(avr, output ? &output_buffer : NULL,
- current_buffer);
-}
-
-int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
-{
- if (avresample_is_open(avr)) {
- avresample_close(avr);
- }
-
- if (in) {
- avr->in_channel_layout = in->channel_layout;
- avr->in_sample_rate = in->sample_rate;
- avr->in_sample_fmt = in->format;
- }
-
- if (out) {
- avr->out_channel_layout = out->channel_layout;
- avr->out_sample_rate = out->sample_rate;
- avr->out_sample_fmt = out->format;
- }
-
- return 0;
-}
-
-static int config_changed(AVAudioResampleContext *avr,
- AVFrame *out, AVFrame *in)
-{
- int ret = 0;
-
- if (in) {
- if (avr->in_channel_layout != in->channel_layout ||
- avr->in_sample_rate != in->sample_rate ||
- avr->in_sample_fmt != in->format) {
- ret |= AVERROR_INPUT_CHANGED;
- }
- }
-
- if (out) {
- if (avr->out_channel_layout != out->channel_layout ||
- avr->out_sample_rate != out->sample_rate ||
- avr->out_sample_fmt != out->format) {
- ret |= AVERROR_OUTPUT_CHANGED;
- }
- }
-
- return ret;
-}
-
-static inline int convert_frame(AVAudioResampleContext *avr,
- AVFrame *out, AVFrame *in)
-{
- int ret;
- uint8_t **out_data = NULL, **in_data = NULL;
- int out_linesize = 0, in_linesize = 0;
- int out_nb_samples = 0, in_nb_samples = 0;
-
- if (out) {
- out_data = out->extended_data;
- out_linesize = out->linesize[0];
- out_nb_samples = out->nb_samples;
- }
-
- if (in) {
- in_data = in->extended_data;
- in_linesize = in->linesize[0];
- in_nb_samples = in->nb_samples;
- }
-
- ret = avresample_convert(avr, out_data, out_linesize,
- out_nb_samples,
- in_data, in_linesize,
- in_nb_samples);
-
- if (ret < 0) {
- if (out)
- out->nb_samples = 0;
- return ret;
- }
-
- if (out)
- out->nb_samples = ret;
-
- return 0;
-}
-
-static inline int available_samples(AVFrame *out)
-{
- int samples;
- int bytes_per_sample = av_get_bytes_per_sample(out->format);
- if (!bytes_per_sample)
- return AVERROR(EINVAL);
-
- samples = out->linesize[0] / bytes_per_sample;
- if (av_sample_fmt_is_planar(out->format)) {
- return samples;
- } else {
- int channels = av_get_channel_layout_nb_channels(out->channel_layout);
- return samples / channels;
- }
-}
-
-int avresample_convert_frame(AVAudioResampleContext *avr,
- AVFrame *out, AVFrame *in)
-{
- int ret, setup = 0;
-
- if (!avresample_is_open(avr)) {
- if ((ret = avresample_config(avr, out, in)) < 0)
- return ret;
- if ((ret = avresample_open(avr)) < 0)
- return ret;
- setup = 1;
- } else {
- // return as is or reconfigure for input changes?
- if ((ret = config_changed(avr, out, in)))
- return ret;
- }
-
- if (out) {
- if (!out->linesize[0]) {
- out->nb_samples = avresample_get_out_samples(avr, in->nb_samples);
- if ((ret = av_frame_get_buffer(out, 0)) < 0) {
- if (setup)
- avresample_close(avr);
- return ret;
- }
- } else {
- if (!out->nb_samples)
- out->nb_samples = available_samples(out);
- }
- }
-
- return convert_frame(avr, out, in);
-}
-
-int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
- int stride)
-{
- int in_channels, out_channels, i, o;
-
- if (avr->am)
- return ff_audio_mix_get_matrix(avr->am, matrix, stride);
-
- in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
- out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
-
- if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
- out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
- return AVERROR(EINVAL);
- }
-
- if (!avr->mix_matrix) {
- av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
- return AVERROR(EINVAL);
- }
-
- for (o = 0; o < out_channels; o++)
- for (i = 0; i < in_channels; i++)
- matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
-
- return 0;
-}
-
-int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
- int stride)
-{
- int in_channels, out_channels, i, o;
-
- if (avr->am)
- return ff_audio_mix_set_matrix(avr->am, matrix, stride);
-
- in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
- out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
-
- if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
- out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
- return AVERROR(EINVAL);
- }
-
- if (avr->mix_matrix)
- av_freep(&avr->mix_matrix);
- avr->mix_matrix = av_malloc(in_channels * out_channels *
- sizeof(*avr->mix_matrix));
- if (!avr->mix_matrix)
- return AVERROR(ENOMEM);
-
- for (o = 0; o < out_channels; o++)
- for (i = 0; i < in_channels; i++)
- avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i];
-
- return 0;
-}
-
-int avresample_set_channel_mapping(AVAudioResampleContext *avr,
- const int *channel_map)
-{
- ChannelMapInfo *info = &avr->ch_map_info;
- int in_channels, ch, i;
-
- in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
- if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) {
- av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
- return AVERROR(EINVAL);
- }
-
- memset(info, 0, sizeof(*info));
- memset(info->input_map, -1, sizeof(info->input_map));
-
- for (ch = 0; ch < in_channels; ch++) {
- if (channel_map[ch] >= in_channels) {
- av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
- return AVERROR(EINVAL);
- }
- if (channel_map[ch] < 0) {
- info->channel_zero[ch] = 1;
- info->channel_map[ch] = -1;
- info->do_zero = 1;
- } else if (info->input_map[channel_map[ch]] >= 0) {
- info->channel_copy[ch] = info->input_map[channel_map[ch]];
- info->channel_map[ch] = -1;
- info->do_copy = 1;
- } else {
- info->channel_map[ch] = channel_map[ch];
- info->input_map[channel_map[ch]] = ch;
- info->do_remap = 1;
- }
- }
- /* Fill-in unmapped input channels with unmapped output channels.
- This is used when remapping during conversion from interleaved to
- planar format. */
- for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
- while (ch < in_channels && info->input_map[ch] >= 0)
- ch++;
- while (i < in_channels && info->channel_map[i] >= 0)
- i++;
- if (ch >= in_channels || i >= in_channels)
- break;
- info->input_map[ch] = i;
- }
-
- avr->use_channel_map = 1;
- return 0;
-}
-
-int avresample_available(AVAudioResampleContext *avr)
-{
- return av_audio_fifo_size(avr->out_fifo);
-}
-
-int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
-{
- int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;
-
- if (avr->resample_needed) {
- samples = av_rescale_rnd(samples,
- avr->out_sample_rate,
- avr->in_sample_rate,
- AV_ROUND_UP);
- }
-
- samples += avresample_available(avr);
-
- if (samples > INT_MAX)
- return AVERROR(EINVAL);
-
- return samples;
-}
-
-int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
-{
- if (!output)
- return av_audio_fifo_drain(avr->out_fifo, nb_samples);
- return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
-}
-
-unsigned avresample_version(void)
-{
- return LIBAVRESAMPLE_VERSION_INT;
-}
-
-const char *avresample_license(void)
-{
-#define LICENSE_PREFIX "libavresample license: "
- return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
-}
-
-const char *avresample_configuration(void)
-{
- return FFMPEG_CONFIGURATION;
-}
diff --git a/libavresample/version.h b/libavresample/version.h
deleted file mode 100644
index d5d3ea82b1..0000000000
--- a/libavresample/version.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * 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 AVRESAMPLE_VERSION_H
-#define AVRESAMPLE_VERSION_H
-
-/**
- * @file
- * @ingroup lavr
- * Libavresample version macros.
- */
-
-#include "libavutil/version.h"
-
-#define LIBAVRESAMPLE_VERSION_MAJOR 4
-#define LIBAVRESAMPLE_VERSION_MINOR 0
-#define LIBAVRESAMPLE_VERSION_MICRO 0
-
-#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
- LIBAVRESAMPLE_VERSION_MINOR, \
- LIBAVRESAMPLE_VERSION_MICRO)
-#define LIBAVRESAMPLE_VERSION AV_VERSION(LIBAVRESAMPLE_VERSION_MAJOR, \
- LIBAVRESAMPLE_VERSION_MINOR, \
- LIBAVRESAMPLE_VERSION_MICRO)
-#define LIBAVRESAMPLE_BUILD LIBAVRESAMPLE_VERSION_INT
-
-#define LIBAVRESAMPLE_IDENT "Lavr" AV_STRINGIFY(LIBAVRESAMPLE_VERSION)
-
-/**
- * FF_API_* defines may be placed below to indicate public API that will be
- * dropped at a future version bump. The defines themselves are not part of
- * the public API and may change, break or disappear at any time.
- */
-
-#endif /* AVRESAMPLE_VERSION_H */
diff --git a/libavresample/x86/Makefile b/libavresample/x86/Makefile
deleted file mode 100644
index 55b709ce36..0000000000
--- a/libavresample/x86/Makefile
+++ /dev/null
@@ -1,9 +0,0 @@
-OBJS += x86/audio_convert_init.o \
- x86/audio_mix_init.o \
- x86/dither_init.o \
-
-OBJS-$(CONFIG_XMM_CLOBBER_TEST) += x86/w64xmmtest.o
-
-X86ASM-OBJS += x86/audio_convert.o \
- x86/audio_mix.o \
- x86/dither.o \
diff --git a/libavresample/x86/audio_convert.asm b/libavresample/x86/audio_convert.asm
deleted file mode 100644
index c6a5015282..0000000000
--- a/libavresample/x86/audio_convert.asm
+++ /dev/null
@@ -1,1261 +0,0 @@
-;******************************************************************************
-;* x86 optimized Format Conversion Utils
-;* Copyright (c) 2008 Loren Merritt
-;* Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
-;*
-;* 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 "libavutil/x86/x86util.asm"
-%include "util.asm"
-
-SECTION_RODATA 32
-
-pf_s32_inv_scale: times 8 dd 0x30000000
-pf_s32_scale: times 8 dd 0x4f000000
-pf_s32_clip: times 8 dd 0x4effffff
-pf_s16_inv_scale: times 4 dd 0x38000000
-pf_s16_scale: times 4 dd 0x47000000
-pb_shuf_unpack_even: db -1, -1, 0, 1, -1, -1, 2, 3, -1, -1, 8, 9, -1, -1, 10, 11
-pb_shuf_unpack_odd: db -1, -1, 4, 5, -1, -1, 6, 7, -1, -1, 12, 13, -1, -1, 14, 15
-pb_interleave_words: SHUFFLE_MASK_W 0, 4, 1, 5, 2, 6, 3, 7
-pb_deinterleave_words: SHUFFLE_MASK_W 0, 2, 4, 6, 1, 3, 5, 7
-pw_zero_even: times 4 dw 0x0000, 0xffff
-
-SECTION .text
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_s32(int32_t *dst, const int16_t *src, int len);
-;------------------------------------------------------------------------------
-
-INIT_XMM sse2
-cglobal conv_s16_to_s32, 3,3,3, dst, src, len
- lea lenq, [2*lend]
- lea dstq, [dstq+2*lenq]
- add srcq, lenq
- neg lenq
-.loop:
- mova m2, [srcq+lenq]
- pxor m0, m0
- pxor m1, m1
- punpcklwd m0, m2
- punpckhwd m1, m2
- mova [dstq+2*lenq ], m0
- mova [dstq+2*lenq+mmsize], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_flt(float *dst, const int16_t *src, int len);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16_TO_FLT 0
-cglobal conv_s16_to_flt, 3,3,3, dst, src, len
- lea lenq, [2*lend]
- add srcq, lenq
- lea dstq, [dstq + 2*lenq]
- neg lenq
- mova m2, [pf_s16_inv_scale]
- ALIGN 16
-.loop:
- mova m0, [srcq+lenq]
- S16_TO_S32_SX 0, 1
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- mulps m0, m2
- mulps m1, m2
- mova [dstq+2*lenq ], m0
- mova [dstq+2*lenq+mmsize], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16_TO_FLT
-INIT_XMM sse4
-CONV_S16_TO_FLT
-
-;------------------------------------------------------------------------------
-; void ff_conv_s32_to_s16(int16_t *dst, const int32_t *src, int len);
-;------------------------------------------------------------------------------
-
-%macro CONV_S32_TO_S16 0
-cglobal conv_s32_to_s16, 3,3,4, dst, src, len
- lea lenq, [2*lend]
- lea srcq, [srcq+2*lenq]
- add dstq, lenq
- neg lenq
-.loop:
- mova m0, [srcq+2*lenq ]
- mova m1, [srcq+2*lenq+ mmsize]
- mova m2, [srcq+2*lenq+2*mmsize]
- mova m3, [srcq+2*lenq+3*mmsize]
- psrad m0, 16
- psrad m1, 16
- psrad m2, 16
- psrad m3, 16
- packssdw m0, m1
- packssdw m2, m3
- mova [dstq+lenq ], m0
- mova [dstq+lenq+mmsize], m2
- add lenq, mmsize*2
- jl .loop
-%if mmsize == 8
- emms
- RET
-%else
- REP_RET
-%endif
-%endmacro
-
-INIT_MMX mmx
-CONV_S32_TO_S16
-INIT_XMM sse2
-CONV_S32_TO_S16
-
-;------------------------------------------------------------------------------
-; void ff_conv_s32_to_flt(float *dst, const int32_t *src, int len);
-;------------------------------------------------------------------------------
-
-%macro CONV_S32_TO_FLT 0
-cglobal conv_s32_to_flt, 3,3,3, dst, src, len
- lea lenq, [4*lend]
- add srcq, lenq
- add dstq, lenq
- neg lenq
- mova m0, [pf_s32_inv_scale]
- ALIGN 16
-.loop:
- cvtdq2ps m1, [srcq+lenq ]
- cvtdq2ps m2, [srcq+lenq+mmsize]
- mulps m1, m1, m0
- mulps m2, m2, m0
- mova [dstq+lenq ], m1
- mova [dstq+lenq+mmsize], m2
- add lenq, mmsize*2
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S32_TO_FLT
-%if HAVE_AVX_EXTERNAL
-INIT_YMM avx
-CONV_S32_TO_FLT
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_s16(int16_t *dst, const float *src, int len);
-;------------------------------------------------------------------------------
-
-INIT_XMM sse2
-cglobal conv_flt_to_s16, 3,3,5, dst, src, len
- lea lenq, [2*lend]
- lea srcq, [srcq+2*lenq]
- add dstq, lenq
- neg lenq
- mova m4, [pf_s16_scale]
-.loop:
- mova m0, [srcq+2*lenq ]
- mova m1, [srcq+2*lenq+1*mmsize]
- mova m2, [srcq+2*lenq+2*mmsize]
- mova m3, [srcq+2*lenq+3*mmsize]
- mulps m0, m4
- mulps m1, m4
- mulps m2, m4
- mulps m3, m4
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- packssdw m0, m1
- packssdw m2, m3
- mova [dstq+lenq ], m0
- mova [dstq+lenq+mmsize], m2
- add lenq, mmsize*2
- jl .loop
- REP_RET
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_s32(int32_t *dst, const float *src, int len);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLT_TO_S32 0
-cglobal conv_flt_to_s32, 3,3,6, dst, src, len
- lea lenq, [lend*4]
- add srcq, lenq
- add dstq, lenq
- neg lenq
- mova m4, [pf_s32_scale]
- mova m5, [pf_s32_clip]
-.loop:
- mulps m0, m4, [srcq+lenq ]
- mulps m1, m4, [srcq+lenq+1*mmsize]
- mulps m2, m4, [srcq+lenq+2*mmsize]
- mulps m3, m4, [srcq+lenq+3*mmsize]
- minps m0, m0, m5
- minps m1, m1, m5
- minps m2, m2, m5
- minps m3, m3, m5
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- mova [dstq+lenq ], m0
- mova [dstq+lenq+1*mmsize], m1
- mova [dstq+lenq+2*mmsize], m2
- mova [dstq+lenq+3*mmsize], m3
- add lenq, mmsize*4
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_FLT_TO_S32
-%if HAVE_AVX_EXTERNAL
-INIT_YMM avx
-CONV_FLT_TO_S32
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16p_to_s16_2ch(int16_t *dst, int16_t *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16P_TO_S16_2CH 0
-cglobal conv_s16p_to_s16_2ch, 3,4,5, dst, src0, len, src1
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q ]
- lea lenq, [2*lend]
- add src0q, lenq
- add src1q, lenq
- lea dstq, [dstq+2*lenq]
- neg lenq
-.loop:
- mova m0, [src0q+lenq ]
- mova m1, [src1q+lenq ]
- mova m2, [src0q+lenq+mmsize]
- mova m3, [src1q+lenq+mmsize]
- SBUTTERFLY2 wd, 0, 1, 4
- SBUTTERFLY2 wd, 2, 3, 4
- mova [dstq+2*lenq+0*mmsize], m0
- mova [dstq+2*lenq+1*mmsize], m1
- mova [dstq+2*lenq+2*mmsize], m2
- mova [dstq+2*lenq+3*mmsize], m3
- add lenq, 2*mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16P_TO_S16_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16P_TO_S16_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16p_to_s16_6ch(int16_t *dst, int16_t *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-;------------------------------------------------------------------------------
-; NOTE: In the 6-channel functions, len could be used as an index on x86-64
-; instead of just a counter, which would avoid incrementing the
-; pointers, but the extra complexity and amount of code is not worth
-; the small gain. On x86-32 there are not enough registers to use len
-; as an index without keeping two of the pointers on the stack and
-; loading them in each iteration.
-;------------------------------------------------------------------------------
-
-%macro CONV_S16P_TO_S16_6CH 0
-%if ARCH_X86_64
-cglobal conv_s16p_to_s16_6ch, 3,8,7, dst, src0, len, src1, src2, src3, src4, src5
-%else
-cglobal conv_s16p_to_s16_6ch, 2,7,7, dst, src0, src1, src2, src3, src4, src5
-%define lend dword r2m
-%endif
- mov src1q, [src0q+1*gprsize]
- mov src2q, [src0q+2*gprsize]
- mov src3q, [src0q+3*gprsize]
- mov src4q, [src0q+4*gprsize]
- mov src5q, [src0q+5*gprsize]
- mov src0q, [src0q]
- sub src1q, src0q
- sub src2q, src0q
- sub src3q, src0q
- sub src4q, src0q
- sub src5q, src0q
-.loop:
-%if cpuflag(sse2slow)
- movq m0, [src0q ] ; m0 = 0, 6, 12, 18, x, x, x, x
- movq m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x
- movq m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x
- movq m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x
- movq m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x
- movq m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x
- ; unpack words:
- punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
- punpcklwd m2, m3 ; m2 = 4, 5, 10, 11, 16, 17, 22, 23
- punpcklwd m4, m5 ; m4 = 2, 3, 8, 9, 14, 15, 20, 21
- ; blend dwords
- shufps m1, m0, m2, q2020 ; m1 = 0, 1, 12, 13, 2, 3, 14, 15
- shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
- shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
- ; shuffle dwords
- pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19
- pshufd m1, m1, q3120 ; m1 = 0, 1, 2, 3, 12, 13, 14, 15
- pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
- movq [dstq+0*mmsize/2], m1
- movq [dstq+1*mmsize/2], m0
- movq [dstq+2*mmsize/2], m2
- movhps [dstq+3*mmsize/2], m1
- movhps [dstq+4*mmsize/2], m0
- movhps [dstq+5*mmsize/2], m2
- add src0q, mmsize/2
- add dstq, mmsize*3
- sub lend, mmsize/4
-%else
- mova m0, [src0q ] ; m0 = 0, 6, 12, 18, 24, 30, 36, 42
- mova m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, 25, 31, 37, 43
- mova m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, 26, 32, 38, 44
- mova m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, 27, 33, 39, 45
- mova m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, 28, 34, 40, 46
- mova m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, 29, 35, 41, 47
- ; unpack words:
- SBUTTERFLY2 wd, 0, 1, 6 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
- ; m1 = 24, 25, 30, 31, 36, 37, 42, 43
- SBUTTERFLY2 wd, 2, 3, 6 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
- ; m3 = 26, 27, 32, 33, 38, 39, 44, 45
- SBUTTERFLY2 wd, 4, 5, 6 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23
- ; m5 = 28, 29, 34, 35, 40, 41, 46, 47
- ; blend dwords
- shufps m6, m0, m2, q2020 ; m6 = 0, 1, 12, 13, 2, 3, 14, 15
- shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
- shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
- SWAP 4,6 ; m4 = 0, 1, 12, 13, 2, 3, 14, 15
- shufps m6, m1, m3, q2020 ; m6 = 24, 25, 36, 37, 26, 27, 38, 39
- shufps m1, m5, q2031 ; m1 = 30, 31, 42, 43, 28, 29, 40, 41
- shufps m3, m5, q3131 ; m3 = 32, 33, 44, 45, 34, 35, 46, 47
- SWAP 5,6 ; m5 = 24, 25, 36, 37, 26, 27, 38, 39
- ; shuffle dwords
- pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19
- pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
- pshufd m4, m4, q3120 ; m4 = 0, 1, 2, 3, 12, 13, 14, 15
- pshufd m1, m1, q1302 ; m1 = 28, 29, 30, 31, 40, 41, 42, 43
- pshufd m3, m3, q3120 ; m3 = 32, 33, 34, 35, 44, 45, 46, 47
- pshufd m5, m5, q3120 ; m5 = 24, 25, 26, 27, 36, 37, 38, 39
- ; shuffle qwords
- punpcklqdq m6, m4, m0 ; m6 = 0, 1, 2, 3, 4, 5, 6, 7
- punpckhqdq m0, m2 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23
- shufps m2, m4, q3210 ; m2 = 8, 9, 10, 11, 12, 13, 14, 15
- SWAP 4,6 ; m4 = 0, 1, 2, 3, 4, 5, 6, 7
- punpcklqdq m6, m5, m1 ; m6 = 24, 25, 26, 27, 28, 29, 30, 31
- punpckhqdq m1, m3 ; m1 = 40, 41, 42, 43, 44, 45, 46, 47
- shufps m3, m5, q3210 ; m3 = 32, 33, 34, 35, 36, 37, 38, 39
- SWAP 5,6 ; m5 = 24, 25, 26, 27, 28, 29, 30, 31
- mova [dstq+0*mmsize], m4
- mova [dstq+1*mmsize], m2
- mova [dstq+2*mmsize], m0
- mova [dstq+3*mmsize], m5
- mova [dstq+4*mmsize], m3
- mova [dstq+5*mmsize], m1
- add src0q, mmsize
- add dstq, mmsize*6
- sub lend, mmsize/2
-%endif
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16P_TO_S16_6CH
-INIT_XMM sse2slow
-CONV_S16P_TO_S16_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16P_TO_S16_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16p_to_flt_2ch(float *dst, int16_t *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16P_TO_FLT_2CH 0
-cglobal conv_s16p_to_flt_2ch, 3,4,6, dst, src0, len, src1
- lea lenq, [2*lend]
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q ]
- lea dstq, [dstq+4*lenq]
- add src0q, lenq
- add src1q, lenq
- neg lenq
- mova m5, [pf_s32_inv_scale]
-.loop:
- mova m2, [src0q+lenq] ; m2 = 0, 2, 4, 6, 8, 10, 12, 14
- mova m4, [src1q+lenq] ; m4 = 1, 3, 5, 7, 9, 11, 13, 15
- SBUTTERFLY2 wd, 2, 4, 3 ; m2 = 0, 1, 2, 3, 4, 5, 6, 7
- ; m4 = 8, 9, 10, 11, 12, 13, 14, 15
- pxor m3, m3
- punpcklwd m0, m3, m2 ; m0 = 0, 1, 2, 3
- punpckhwd m1, m3, m2 ; m1 = 4, 5, 6, 7
- punpcklwd m2, m3, m4 ; m2 = 8, 9, 10, 11
- punpckhwd m3, m4 ; m3 = 12, 13, 14, 15
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- cvtdq2ps m2, m2
- cvtdq2ps m3, m3
- mulps m0, m5
- mulps m1, m5
- mulps m2, m5
- mulps m3, m5
- mova [dstq+4*lenq ], m0
- mova [dstq+4*lenq+ mmsize], m1
- mova [dstq+4*lenq+2*mmsize], m2
- mova [dstq+4*lenq+3*mmsize], m3
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16P_TO_FLT_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16P_TO_FLT_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16p_to_flt_6ch(float *dst, int16_t *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16P_TO_FLT_6CH 0
-%if ARCH_X86_64
-cglobal conv_s16p_to_flt_6ch, 3,8,8, dst, src, len, src1, src2, src3, src4, src5
-%else
-cglobal conv_s16p_to_flt_6ch, 2,7,8, dst, src, src1, src2, src3, src4, src5
-%define lend dword r2m
-%endif
- mov src1q, [srcq+1*gprsize]
- mov src2q, [srcq+2*gprsize]
- mov src3q, [srcq+3*gprsize]
- mov src4q, [srcq+4*gprsize]
- mov src5q, [srcq+5*gprsize]
- mov srcq, [srcq]
- sub src1q, srcq
- sub src2q, srcq
- sub src3q, srcq
- sub src4q, srcq
- sub src5q, srcq
- mova m7, [pf_s32_inv_scale]
-%if cpuflag(ssse3)
- %define unpack_even m6
- mova m6, [pb_shuf_unpack_even]
-%if ARCH_X86_64
- %define unpack_odd m8
- mova m8, [pb_shuf_unpack_odd]
-%else
- %define unpack_odd [pb_shuf_unpack_odd]
-%endif
-%endif
-.loop:
- movq m0, [srcq ] ; m0 = 0, 6, 12, 18, x, x, x, x
- movq m1, [srcq+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x
- movq m2, [srcq+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x
- movq m3, [srcq+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x
- movq m4, [srcq+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x
- movq m5, [srcq+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x
- ; unpack words:
- punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
- punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
- punpcklwd m4, m5 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23
- ; blend dwords
- shufps m1, m4, m0, q3120 ; m1 = 4, 5, 16, 17, 6, 7, 18, 19
- shufps m0, m2, q2020 ; m0 = 0, 1, 12, 13, 2, 3, 14, 15
- shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
-%if cpuflag(ssse3)
- pshufb m3, m0, unpack_odd ; m3 = 12, 13, 14, 15
- pshufb m0, unpack_even ; m0 = 0, 1, 2, 3
- pshufb m4, m1, unpack_odd ; m4 = 16, 17, 18, 19
- pshufb m1, unpack_even ; m1 = 4, 5, 6, 7
- pshufb m5, m2, unpack_odd ; m5 = 20, 21, 22, 23
- pshufb m2, unpack_even ; m2 = 8, 9, 10, 11
-%else
- ; shuffle dwords
- pshufd m0, m0, q3120 ; m0 = 0, 1, 2, 3, 12, 13, 14, 15
- pshufd m1, m1, q3120 ; m1 = 4, 5, 6, 7, 16, 17, 18, 19
- pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23
- pxor m6, m6 ; convert s16 in m0-m2 to s32 in m0-m5
- punpcklwd m3, m6, m0 ; m3 = 0, 1, 2, 3
- punpckhwd m4, m6, m0 ; m4 = 12, 13, 14, 15
- punpcklwd m0, m6, m1 ; m0 = 4, 5, 6, 7
- punpckhwd m5, m6, m1 ; m5 = 16, 17, 18, 19
- punpcklwd m1, m6, m2 ; m1 = 8, 9, 10, 11
- punpckhwd m6, m2 ; m6 = 20, 21, 22, 23
- SWAP 6,2,1,0,3,4,5 ; swap registers 3,0,1,4,5,6 to 0,1,2,3,4,5
-%endif
- cvtdq2ps m0, m0 ; convert s32 to float
- cvtdq2ps m1, m1
- cvtdq2ps m2, m2
- cvtdq2ps m3, m3
- cvtdq2ps m4, m4
- cvtdq2ps m5, m5
- mulps m0, m7 ; scale float from s32 range to [-1.0,1.0]
- mulps m1, m7
- mulps m2, m7
- mulps m3, m7
- mulps m4, m7
- mulps m5, m7
- mova [dstq ], m0
- mova [dstq+ mmsize], m1
- mova [dstq+2*mmsize], m2
- mova [dstq+3*mmsize], m3
- mova [dstq+4*mmsize], m4
- mova [dstq+5*mmsize], m5
- add srcq, mmsize/2
- add dstq, mmsize*6
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16P_TO_FLT_6CH
-INIT_XMM ssse3
-CONV_S16P_TO_FLT_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16P_TO_FLT_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_fltp_to_s16_2ch(int16_t *dst, float *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLTP_TO_S16_2CH 0
-cglobal conv_fltp_to_s16_2ch, 3,4,3, dst, src0, len, src1
- lea lenq, [4*lend]
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q ]
- add dstq, lenq
- add src0q, lenq
- add src1q, lenq
- neg lenq
- mova m2, [pf_s16_scale]
-%if cpuflag(ssse3)
- mova m3, [pb_interleave_words]
-%endif
-.loop:
- mulps m0, m2, [src0q+lenq] ; m0 = 0, 2, 4, 6
- mulps m1, m2, [src1q+lenq] ; m1 = 1, 3, 5, 7
- cvtps2dq m0, m0
- cvtps2dq m1, m1
-%if cpuflag(ssse3)
- packssdw m0, m1 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7
- pshufb m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
-%else
- packssdw m0, m0 ; m0 = 0, 2, 4, 6, x, x, x, x
- packssdw m1, m1 ; m1 = 1, 3, 5, 7, x, x, x, x
- punpcklwd m0, m1 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
-%endif
- mova [dstq+lenq], m0
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_FLTP_TO_S16_2CH
-INIT_XMM ssse3
-CONV_FLTP_TO_S16_2CH
-
-;------------------------------------------------------------------------------
-; void ff_conv_fltp_to_s16_6ch(int16_t *dst, float *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLTP_TO_S16_6CH 0
-%if ARCH_X86_64
-cglobal conv_fltp_to_s16_6ch, 3,8,7, dst, src, len, src1, src2, src3, src4, src5
-%else
-cglobal conv_fltp_to_s16_6ch, 2,7,7, dst, src, src1, src2, src3, src4, src5
-%define lend dword r2m
-%endif
- mov src1q, [srcq+1*gprsize]
- mov src2q, [srcq+2*gprsize]
- mov src3q, [srcq+3*gprsize]
- mov src4q, [srcq+4*gprsize]
- mov src5q, [srcq+5*gprsize]
- mov srcq, [srcq]
- sub src1q, srcq
- sub src2q, srcq
- sub src3q, srcq
- sub src4q, srcq
- sub src5q, srcq
- movaps xmm6, [pf_s16_scale]
-.loop:
-%if cpuflag(sse2)
- mulps m0, m6, [srcq ]
- mulps m1, m6, [srcq+src1q]
- mulps m2, m6, [srcq+src2q]
- mulps m3, m6, [srcq+src3q]
- mulps m4, m6, [srcq+src4q]
- mulps m5, m6, [srcq+src5q]
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- cvtps2dq m4, m4
- cvtps2dq m5, m5
- packssdw m0, m3 ; m0 = 0, 6, 12, 18, 3, 9, 15, 21
- packssdw m1, m4 ; m1 = 1, 7, 13, 19, 4, 10, 16, 22
- packssdw m2, m5 ; m2 = 2, 8, 14, 20, 5, 11, 17, 23
- ; unpack words:
- movhlps m3, m0 ; m3 = 3, 9, 15, 21, x, x, x, x
- punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19
- punpckhwd m1, m2 ; m1 = 4, 5, 10, 11, 16, 17, 22, 23
- punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21
- ; blend dwords:
- shufps m3, m0, m2, q2020 ; m3 = 0, 1, 12, 13, 2, 3, 14, 15
- shufps m0, m1, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17
- shufps m2, m1, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23
- ; shuffle dwords:
- shufps m1, m2, m3, q3120 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
- shufps m3, m0, q0220 ; m3 = 0, 1, 2, 3, 4, 5, 6, 7
- shufps m0, m2, q3113 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23
- mova [dstq+0*mmsize], m3
- mova [dstq+1*mmsize], m1
- mova [dstq+2*mmsize], m0
-%else ; sse
- movlps xmm0, [srcq ]
- movlps xmm1, [srcq+src1q]
- movlps xmm2, [srcq+src2q]
- movlps xmm3, [srcq+src3q]
- movlps xmm4, [srcq+src4q]
- movlps xmm5, [srcq+src5q]
- mulps xmm0, xmm6
- mulps xmm1, xmm6
- mulps xmm2, xmm6
- mulps xmm3, xmm6
- mulps xmm4, xmm6
- mulps xmm5, xmm6
- cvtps2pi mm0, xmm0
- cvtps2pi mm1, xmm1
- cvtps2pi mm2, xmm2
- cvtps2pi mm3, xmm3
- cvtps2pi mm4, xmm4
- cvtps2pi mm5, xmm5
- packssdw mm0, mm3 ; m0 = 0, 6, 3, 9
- packssdw mm1, mm4 ; m1 = 1, 7, 4, 10
- packssdw mm2, mm5 ; m2 = 2, 8, 5, 11
- ; unpack words
- pshufw mm3, mm0, q1032 ; m3 = 3, 9, 0, 6
- punpcklwd mm0, mm1 ; m0 = 0, 1, 6, 7
- punpckhwd mm1, mm2 ; m1 = 4, 5, 10, 11
- punpcklwd mm2, mm3 ; m2 = 2, 3, 8, 9
- ; unpack dwords
- pshufw mm3, mm0, q1032 ; m3 = 6, 7, 0, 1
- punpckldq mm0, mm2 ; m0 = 0, 1, 2, 3 (final)
- punpckhdq mm2, mm1 ; m2 = 8, 9, 10, 11 (final)
- punpckldq mm1, mm3 ; m1 = 4, 5, 6, 7 (final)
- mova [dstq+0*mmsize], mm0
- mova [dstq+1*mmsize], mm1
- mova [dstq+2*mmsize], mm2
-%endif
- add srcq, mmsize
- add dstq, mmsize*3
- sub lend, mmsize/4
- jg .loop
-%if mmsize == 8
- emms
- RET
-%else
- REP_RET
-%endif
-%endmacro
-
-INIT_MMX sse
-CONV_FLTP_TO_S16_6CH
-INIT_XMM sse2
-CONV_FLTP_TO_S16_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLTP_TO_S16_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_fltp_to_flt_2ch(float *dst, float *const *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLTP_TO_FLT_2CH 0
-cglobal conv_fltp_to_flt_2ch, 3,4,5, dst, src0, len, src1
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q]
- lea lenq, [4*lend]
- add src0q, lenq
- add src1q, lenq
- lea dstq, [dstq+2*lenq]
- neg lenq
-.loop:
- mova m0, [src0q+lenq ]
- mova m1, [src1q+lenq ]
- mova m2, [src0q+lenq+mmsize]
- mova m3, [src1q+lenq+mmsize]
- SBUTTERFLYPS 0, 1, 4
- SBUTTERFLYPS 2, 3, 4
- mova [dstq+2*lenq+0*mmsize], m0
- mova [dstq+2*lenq+1*mmsize], m1
- mova [dstq+2*lenq+2*mmsize], m2
- mova [dstq+2*lenq+3*mmsize], m3
- add lenq, 2*mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse
-CONV_FLTP_TO_FLT_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLTP_TO_FLT_2CH
-%endif
-
-;-----------------------------------------------------------------------------
-; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len,
-; int channels);
-;-----------------------------------------------------------------------------
-
-%macro CONV_FLTP_TO_FLT_6CH 0
-cglobal conv_fltp_to_flt_6ch, 2,8,7, dst, src, src1, src2, src3, src4, src5, len
-%if ARCH_X86_64
- mov lend, r2d
-%else
- %define lend dword r2m
-%endif
- mov src1q, [srcq+1*gprsize]
- mov src2q, [srcq+2*gprsize]
- mov src3q, [srcq+3*gprsize]
- mov src4q, [srcq+4*gprsize]
- mov src5q, [srcq+5*gprsize]
- mov srcq, [srcq]
- sub src1q, srcq
- sub src2q, srcq
- sub src3q, srcq
- sub src4q, srcq
- sub src5q, srcq
-.loop:
- mova m0, [srcq ]
- mova m1, [srcq+src1q]
- mova m2, [srcq+src2q]
- mova m3, [srcq+src3q]
- mova m4, [srcq+src4q]
- mova m5, [srcq+src5q]
-%if cpuflag(sse4)
- SBUTTERFLYPS 0, 1, 6
- SBUTTERFLYPS 2, 3, 6
- SBUTTERFLYPS 4, 5, 6
-
- blendps m6, m4, m0, 1100b
- movlhps m0, m2
- movhlps m4, m2
- blendps m2, m5, m1, 1100b
- movlhps m1, m3
- movhlps m5, m3
-
- movaps [dstq ], m0
- movaps [dstq+16], m6
- movaps [dstq+32], m4
- movaps [dstq+48], m1
- movaps [dstq+64], m2
- movaps [dstq+80], m5
-%else ; mmx
- SBUTTERFLY dq, 0, 1, 6
- SBUTTERFLY dq, 2, 3, 6
- SBUTTERFLY dq, 4, 5, 6
-
- movq [dstq ], m0
- movq [dstq+ 8], m2
- movq [dstq+16], m4
- movq [dstq+24], m1
- movq [dstq+32], m3
- movq [dstq+40], m5
-%endif
- add srcq, mmsize
- add dstq, mmsize*6
- sub lend, mmsize/4
- jg .loop
-%if mmsize == 8
- emms
- RET
-%else
- REP_RET
-%endif
-%endmacro
-
-INIT_MMX mmx
-CONV_FLTP_TO_FLT_6CH
-INIT_XMM sse4
-CONV_FLTP_TO_FLT_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLTP_TO_FLT_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_s16p_2ch(int16_t *const *dst, int16_t *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16_TO_S16P_2CH 0
-cglobal conv_s16_to_s16p_2ch, 3,4,4, dst0, src, len, dst1
- lea lenq, [2*lend]
- mov dst1q, [dst0q+gprsize]
- mov dst0q, [dst0q ]
- lea srcq, [srcq+2*lenq]
- add dst0q, lenq
- add dst1q, lenq
- neg lenq
-%if cpuflag(ssse3)
- mova m3, [pb_deinterleave_words]
-%endif
-.loop:
- mova m0, [srcq+2*lenq ] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
- mova m1, [srcq+2*lenq+mmsize] ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
-%if cpuflag(ssse3)
- pshufb m0, m3 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7
- pshufb m1, m3 ; m1 = 8, 10, 12, 14, 9, 11, 13, 15
- SBUTTERFLY2 qdq, 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14
- ; m1 = 1, 3, 5, 7, 9, 11, 13, 15
-%else ; sse2
- pshuflw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 5, 6, 7
- pshufhw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 6, 5, 7
- pshuflw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 13, 14, 15
- pshufhw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 14, 13, 15
- DEINT2_PS 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14
- ; m1 = 1, 3, 5, 7, 9, 11, 13, 15
-%endif
- mova [dst0q+lenq], m0
- mova [dst1q+lenq], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16_TO_S16P_2CH
-INIT_XMM ssse3
-CONV_S16_TO_S16P_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16_TO_S16P_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_s16p_6ch(int16_t *const *dst, int16_t *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16_TO_S16P_6CH 0
-%if ARCH_X86_64
-cglobal conv_s16_to_s16p_6ch, 3,8,5, dst, src, len, dst1, dst2, dst3, dst4, dst5
-%else
-cglobal conv_s16_to_s16p_6ch, 2,7,5, dst, src, dst1, dst2, dst3, dst4, dst5
-%define lend dword r2m
-%endif
- mov dst1q, [dstq+ gprsize]
- mov dst2q, [dstq+2*gprsize]
- mov dst3q, [dstq+3*gprsize]
- mov dst4q, [dstq+4*gprsize]
- mov dst5q, [dstq+5*gprsize]
- mov dstq, [dstq ]
- sub dst1q, dstq
- sub dst2q, dstq
- sub dst3q, dstq
- sub dst4q, dstq
- sub dst5q, dstq
-.loop:
- mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
- mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15
- mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
- PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x
- shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19
- psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
- SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
- ; m1 = 4, 10, 5, 11, x, x, x, x
- SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21
- ; m2 = 16, 22, 17, 23, x, x, x, x
- SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
- ; m3 = 2, 8, 14, 20, 3, 9, 15, 21
- punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23
- movq [dstq ], m0
- movhps [dstq+dst1q], m0
- movq [dstq+dst2q], m3
- movhps [dstq+dst3q], m3
- movq [dstq+dst4q], m1
- movhps [dstq+dst5q], m1
- add srcq, mmsize*3
- add dstq, mmsize/2
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16_TO_S16P_6CH
-INIT_XMM ssse3
-CONV_S16_TO_S16P_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16_TO_S16P_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_fltp_2ch(float *const *dst, int16_t *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16_TO_FLTP_2CH 0
-cglobal conv_s16_to_fltp_2ch, 3,4,5, dst0, src, len, dst1
- lea lenq, [4*lend]
- mov dst1q, [dst0q+gprsize]
- mov dst0q, [dst0q ]
- add srcq, lenq
- add dst0q, lenq
- add dst1q, lenq
- neg lenq
- mova m3, [pf_s32_inv_scale]
- mova m4, [pw_zero_even]
-.loop:
- mova m1, [srcq+lenq]
- pslld m0, m1, 16
- pand m1, m4
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- mulps m0, m0, m3
- mulps m1, m1, m3
- mova [dst0q+lenq], m0
- mova [dst1q+lenq], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16_TO_FLTP_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16_TO_FLTP_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_s16_to_fltp_6ch(float *const *dst, int16_t *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_S16_TO_FLTP_6CH 0
-%if ARCH_X86_64
-cglobal conv_s16_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
-%else
-cglobal conv_s16_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
-%define lend dword r2m
-%endif
- mov dst1q, [dstq+ gprsize]
- mov dst2q, [dstq+2*gprsize]
- mov dst3q, [dstq+3*gprsize]
- mov dst4q, [dstq+4*gprsize]
- mov dst5q, [dstq+5*gprsize]
- mov dstq, [dstq ]
- sub dst1q, dstq
- sub dst2q, dstq
- sub dst3q, dstq
- sub dst4q, dstq
- sub dst5q, dstq
- mova m6, [pf_s16_inv_scale]
-.loop:
- mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
- mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15
- mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
- PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x
- shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19
- psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
- SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
- ; m1 = 4, 10, 5, 11, x, x, x, x
- SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21
- ; m2 = 16, 22, 17, 23, x, x, x, x
- SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
- ; m3 = 2, 8, 14, 20, 3, 9, 15, 21
- punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23
- S16_TO_S32_SX 0, 2 ; m0 = 0, 6, 12, 18
- ; m2 = 1, 7, 13, 19
- S16_TO_S32_SX 3, 4 ; m3 = 2, 8, 14, 20
- ; m4 = 3, 9, 15, 21
- S16_TO_S32_SX 1, 5 ; m1 = 4, 10, 16, 22
- ; m5 = 5, 11, 17, 23
- SWAP 1,2,3,4
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- cvtdq2ps m2, m2
- cvtdq2ps m3, m3
- cvtdq2ps m4, m4
- cvtdq2ps m5, m5
- mulps m0, m6
- mulps m1, m6
- mulps m2, m6
- mulps m3, m6
- mulps m4, m6
- mulps m5, m6
- mova [dstq ], m0
- mova [dstq+dst1q], m1
- mova [dstq+dst2q], m2
- mova [dstq+dst3q], m3
- mova [dstq+dst4q], m4
- mova [dstq+dst5q], m5
- add srcq, mmsize*3
- add dstq, mmsize
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_S16_TO_FLTP_6CH
-INIT_XMM ssse3
-CONV_S16_TO_FLTP_6CH
-INIT_XMM sse4
-CONV_S16_TO_FLTP_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_S16_TO_FLTP_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_s16p_2ch(int16_t *const *dst, float *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLT_TO_S16P_2CH 0
-cglobal conv_flt_to_s16p_2ch, 3,4,6, dst0, src, len, dst1
- lea lenq, [2*lend]
- mov dst1q, [dst0q+gprsize]
- mov dst0q, [dst0q ]
- lea srcq, [srcq+4*lenq]
- add dst0q, lenq
- add dst1q, lenq
- neg lenq
- mova m5, [pf_s16_scale]
-.loop:
- mova m0, [srcq+4*lenq ]
- mova m1, [srcq+4*lenq+ mmsize]
- mova m2, [srcq+4*lenq+2*mmsize]
- mova m3, [srcq+4*lenq+3*mmsize]
- DEINT2_PS 0, 1, 4
- DEINT2_PS 2, 3, 4
- mulps m0, m0, m5
- mulps m1, m1, m5
- mulps m2, m2, m5
- mulps m3, m3, m5
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- packssdw m0, m2
- packssdw m1, m3
- mova [dst0q+lenq], m0
- mova [dst1q+lenq], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_FLT_TO_S16P_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLT_TO_S16P_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_s16p_6ch(int16_t *const *dst, float *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLT_TO_S16P_6CH 0
-%if ARCH_X86_64
-cglobal conv_flt_to_s16p_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
-%else
-cglobal conv_flt_to_s16p_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
-%define lend dword r2m
-%endif
- mov dst1q, [dstq+ gprsize]
- mov dst2q, [dstq+2*gprsize]
- mov dst3q, [dstq+3*gprsize]
- mov dst4q, [dstq+4*gprsize]
- mov dst5q, [dstq+5*gprsize]
- mov dstq, [dstq ]
- sub dst1q, dstq
- sub dst2q, dstq
- sub dst3q, dstq
- sub dst4q, dstq
- sub dst5q, dstq
- mova m6, [pf_s16_scale]
-.loop:
- mulps m0, m6, [srcq+0*mmsize]
- mulps m3, m6, [srcq+1*mmsize]
- mulps m1, m6, [srcq+2*mmsize]
- mulps m4, m6, [srcq+3*mmsize]
- mulps m2, m6, [srcq+4*mmsize]
- mulps m5, m6, [srcq+5*mmsize]
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- cvtps2dq m4, m4
- cvtps2dq m5, m5
- packssdw m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7
- packssdw m1, m4 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15
- packssdw m2, m5 ; m2 = 16, 17, 18, 19, 20, 21, 22, 23
- PALIGNR m3, m1, m0, 12, m4 ; m3 = 6, 7, 8, 9, 10, 11, x, x
- shufps m1, m2, q1032 ; m1 = 12, 13, 14, 15, 16, 17, 18, 19
- psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x
- SBUTTERFLY2 wd, 0, 3, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9
- ; m3 = 4, 10, 5, 11, x, x, x, x
- SBUTTERFLY2 wd, 1, 2, 4 ; m1 = 12, 18, 13, 19, 14, 20, 15, 21
- ; m2 = 16, 22, 17, 23, x, x, x, x
- SBUTTERFLY2 dq, 0, 1, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19
- ; m1 = 2, 8, 14, 20, 3, 9, 15, 21
- punpckldq m3, m2 ; m3 = 4, 10, 16, 22, 5, 11, 17, 23
- movq [dstq ], m0
- movhps [dstq+dst1q], m0
- movq [dstq+dst2q], m1
- movhps [dstq+dst3q], m1
- movq [dstq+dst4q], m3
- movhps [dstq+dst5q], m3
- add srcq, mmsize*6
- add dstq, mmsize/2
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_FLT_TO_S16P_6CH
-INIT_XMM ssse3
-CONV_FLT_TO_S16P_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLT_TO_S16P_6CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_fltp_2ch(float *const *dst, float *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLT_TO_FLTP_2CH 0
-cglobal conv_flt_to_fltp_2ch, 3,4,3, dst0, src, len, dst1
- lea lenq, [4*lend]
- mov dst1q, [dst0q+gprsize]
- mov dst0q, [dst0q ]
- lea srcq, [srcq+2*lenq]
- add dst0q, lenq
- add dst1q, lenq
- neg lenq
-.loop:
- mova m0, [srcq+2*lenq ]
- mova m1, [srcq+2*lenq+mmsize]
- DEINT2_PS 0, 1, 2
- mova [dst0q+lenq], m0
- mova [dst1q+lenq], m1
- add lenq, mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse
-CONV_FLT_TO_FLTP_2CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLT_TO_FLTP_2CH
-%endif
-
-;------------------------------------------------------------------------------
-; void ff_conv_flt_to_fltp_6ch(float *const *dst, float *src, int len,
-; int channels);
-;------------------------------------------------------------------------------
-
-%macro CONV_FLT_TO_FLTP_6CH 0
-%if ARCH_X86_64
-cglobal conv_flt_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5
-%else
-cglobal conv_flt_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5
-%define lend dword r2m
-%endif
- mov dst1q, [dstq+ gprsize]
- mov dst2q, [dstq+2*gprsize]
- mov dst3q, [dstq+3*gprsize]
- mov dst4q, [dstq+4*gprsize]
- mov dst5q, [dstq+5*gprsize]
- mov dstq, [dstq ]
- sub dst1q, dstq
- sub dst2q, dstq
- sub dst3q, dstq
- sub dst4q, dstq
- sub dst5q, dstq
-.loop:
- mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3
- mova m1, [srcq+1*mmsize] ; m1 = 4, 5, 6, 7
- mova m2, [srcq+2*mmsize] ; m2 = 8, 9, 10, 11
- mova m3, [srcq+3*mmsize] ; m3 = 12, 13, 14, 15
- mova m4, [srcq+4*mmsize] ; m4 = 16, 17, 18, 19
- mova m5, [srcq+5*mmsize] ; m5 = 20, 21, 22, 23
-
- SBUTTERFLY2 dq, 0, 3, 6 ; m0 = 0, 12, 1, 13
- ; m3 = 2, 14, 3, 15
- SBUTTERFLY2 dq, 1, 4, 6 ; m1 = 4, 16, 5, 17
- ; m4 = 6, 18, 7, 19
- SBUTTERFLY2 dq, 2, 5, 6 ; m2 = 8, 20, 9, 21
- ; m5 = 10, 22, 11, 23
- SBUTTERFLY2 dq, 0, 4, 6 ; m0 = 0, 6, 12, 18
- ; m4 = 1, 7, 13, 19
- SBUTTERFLY2 dq, 3, 2, 6 ; m3 = 2, 8, 14, 20
- ; m2 = 3, 9, 15, 21
- SBUTTERFLY2 dq, 1, 5, 6 ; m1 = 4, 10, 16, 22
- ; m5 = 5, 11, 17, 23
- mova [dstq ], m0
- mova [dstq+dst1q], m4
- mova [dstq+dst2q], m3
- mova [dstq+dst3q], m2
- mova [dstq+dst4q], m1
- mova [dstq+dst5q], m5
- add srcq, mmsize*6
- add dstq, mmsize
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-CONV_FLT_TO_FLTP_6CH
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-CONV_FLT_TO_FLTP_6CH
-%endif
diff --git a/libavresample/x86/audio_convert_init.c b/libavresample/x86/audio_convert_init.c
deleted file mode 100644
index 0af4222bea..0000000000
--- a/libavresample/x86/audio_convert_init.c
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 "config.h"
-#include "libavutil/cpu.h"
-#include "libavutil/x86/cpu.h"
-#include "libavresample/audio_convert.h"
-
-/* flat conversions */
-
-void ff_conv_s16_to_s32_sse2(int16_t *dst, const int32_t *src, int len);
-
-void ff_conv_s16_to_flt_sse2(float *dst, const int16_t *src, int len);
-void ff_conv_s16_to_flt_sse4(float *dst, const int16_t *src, int len);
-
-void ff_conv_s32_to_s16_mmx (int16_t *dst, const int32_t *src, int len);
-void ff_conv_s32_to_s16_sse2(int16_t *dst, const int32_t *src, int len);
-
-void ff_conv_s32_to_flt_sse2(float *dst, const int32_t *src, int len);
-void ff_conv_s32_to_flt_avx (float *dst, const int32_t *src, int len);
-
-void ff_conv_flt_to_s16_sse2(int16_t *dst, const float *src, int len);
-
-void ff_conv_flt_to_s32_sse2(int32_t *dst, const float *src, int len);
-void ff_conv_flt_to_s32_avx (int32_t *dst, const float *src, int len);
-
-/* interleave conversions */
-
-void ff_conv_s16p_to_s16_2ch_sse2(int16_t *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_s16_2ch_avx (int16_t *dst, int16_t *const *src,
- int len, int channels);
-
-void ff_conv_s16p_to_s16_6ch_sse2(int16_t *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_s16_6ch_sse2slow(int16_t *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_s16_6ch_avx (int16_t *dst, int16_t *const *src,
- int len, int channels);
-
-void ff_conv_s16p_to_flt_2ch_sse2(float *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_flt_2ch_avx (float *dst, int16_t *const *src,
- int len, int channels);
-
-void ff_conv_s16p_to_flt_6ch_sse2 (float *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_flt_6ch_ssse3(float *dst, int16_t *const *src,
- int len, int channels);
-void ff_conv_s16p_to_flt_6ch_avx (float *dst, int16_t *const *src,
- int len, int channels);
-
-void ff_conv_fltp_to_s16_2ch_sse2 (int16_t *dst, float *const *src,
- int len, int channels);
-void ff_conv_fltp_to_s16_2ch_ssse3(int16_t *dst, float *const *src,
- int len, int channels);
-
-void ff_conv_fltp_to_s16_6ch_sse (int16_t *dst, float *const *src,
- int len, int channels);
-void ff_conv_fltp_to_s16_6ch_sse2(int16_t *dst, float *const *src,
- int len, int channels);
-void ff_conv_fltp_to_s16_6ch_avx (int16_t *dst, float *const *src,
- int len, int channels);
-
-void ff_conv_fltp_to_flt_2ch_sse(float *dst, float *const *src, int len,
- int channels);
-void ff_conv_fltp_to_flt_2ch_avx(float *dst, float *const *src, int len,
- int channels);
-
-void ff_conv_fltp_to_flt_6ch_mmx (float *dst, float *const *src, int len,
- int channels);
-void ff_conv_fltp_to_flt_6ch_sse4(float *dst, float *const *src, int len,
- int channels);
-void ff_conv_fltp_to_flt_6ch_avx (float *dst, float *const *src, int len,
- int channels);
-
-/* deinterleave conversions */
-
-void ff_conv_s16_to_s16p_2ch_sse2(int16_t *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_s16p_2ch_ssse3(int16_t *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_s16p_2ch_avx (int16_t *const *dst, int16_t *src,
- int len, int channels);
-
-void ff_conv_s16_to_s16p_6ch_sse2 (int16_t *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_s16p_6ch_ssse3(int16_t *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_s16p_6ch_avx (int16_t *const *dst, int16_t *src,
- int len, int channels);
-
-void ff_conv_s16_to_fltp_2ch_sse2(float *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_fltp_2ch_avx (float *const *dst, int16_t *src,
- int len, int channels);
-
-void ff_conv_s16_to_fltp_6ch_sse2 (float *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_fltp_6ch_ssse3(float *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_fltp_6ch_sse4 (float *const *dst, int16_t *src,
- int len, int channels);
-void ff_conv_s16_to_fltp_6ch_avx (float *const *dst, int16_t *src,
- int len, int channels);
-
-void ff_conv_flt_to_s16p_2ch_sse2(int16_t *const *dst, float *src,
- int len, int channels);
-void ff_conv_flt_to_s16p_2ch_avx (int16_t *const *dst, float *src,
- int len, int channels);
-
-void ff_conv_flt_to_s16p_6ch_sse2 (int16_t *const *dst, float *src,
- int len, int channels);
-void ff_conv_flt_to_s16p_6ch_ssse3(int16_t *const *dst, float *src,
- int len, int channels);
-void ff_conv_flt_to_s16p_6ch_avx (int16_t *const *dst, float *src,
- int len, int channels);
-
-void ff_conv_flt_to_fltp_2ch_sse(float *const *dst, float *src, int len,
- int channels);
-void ff_conv_flt_to_fltp_2ch_avx(float *const *dst, float *src, int len,
- int channels);
-
-void ff_conv_flt_to_fltp_6ch_sse2(float *const *dst, float *src, int len,
- int channels);
-void ff_conv_flt_to_fltp_6ch_avx (float *const *dst, float *src, int len,
- int channels);
-
-av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (EXTERNAL_MMX(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
- 0, 1, 8, "MMX", ff_conv_s32_to_s16_mmx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- 6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx);
- }
- if (EXTERNAL_SSE(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 6, 1, 2, "SSE", ff_conv_fltp_to_s16_6ch_sse);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- 2, 16, 8, "SSE", ff_conv_fltp_to_flt_2ch_sse);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
- 2, 16, 4, "SSE", ff_conv_flt_to_fltp_2ch_sse);
- }
- if (EXTERNAL_SSE2(cpu_flags)) {
- if (!(cpu_flags & AV_CPU_FLAG_SSE2SLOW)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
- 0, 16, 16, "SSE2", ff_conv_s32_to_s16_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- 6, 16, 8, "SSE2", ff_conv_s16p_to_s16_6ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 6, 16, 4, "SSE2", ff_conv_fltp_to_s16_6ch_sse2);
- } else {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- 6, 1, 4, "SSE2SLOW", ff_conv_s16p_to_s16_6ch_sse2slow);
- }
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16,
- 0, 16, 8, "SSE2", ff_conv_s16_to_s32_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
- 0, 16, 8, "SSE2", ff_conv_s16_to_flt_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
- 0, 16, 8, "SSE2", ff_conv_s32_to_flt_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
- 0, 16, 16, "SSE2", ff_conv_flt_to_s16_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
- 0, 16, 16, "SSE2", ff_conv_flt_to_s32_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- 2, 16, 16, "SSE2", ff_conv_s16p_to_s16_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
- 2, 16, 8, "SSE2", ff_conv_s16p_to_flt_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
- 6, 16, 4, "SSE2", ff_conv_s16p_to_flt_6ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 2, 16, 4, "SSE2", ff_conv_fltp_to_s16_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 2, 16, 8, "SSE2", ff_conv_s16_to_s16p_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "SSE2", ff_conv_s16_to_s16p_6ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
- 2, 16, 8, "SSE2", ff_conv_s16_to_fltp_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "SSE2", ff_conv_s16_to_fltp_6ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
- 2, 16, 8, "SSE2", ff_conv_flt_to_s16p_2ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
- 6, 16, 4, "SSE2", ff_conv_flt_to_s16p_6ch_sse2);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
- 6, 16, 4, "SSE2", ff_conv_flt_to_fltp_6ch_sse2);
- }
- if (EXTERNAL_SSSE3(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
- 6, 16, 4, "SSSE3", ff_conv_s16p_to_flt_6ch_ssse3);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 2, 16, 4, "SSSE3", ff_conv_fltp_to_s16_2ch_ssse3);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 2, 16, 8, "SSSE3", ff_conv_s16_to_s16p_2ch_ssse3);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "SSSE3", ff_conv_s16_to_s16p_6ch_ssse3);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "SSSE3", ff_conv_s16_to_fltp_6ch_ssse3);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
- 6, 16, 4, "SSSE3", ff_conv_flt_to_s16p_6ch_ssse3);
- }
- if (EXTERNAL_SSE4(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
- 0, 16, 8, "SSE4", ff_conv_s16_to_flt_sse4);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- 6, 16, 4, "SSE4", ff_conv_fltp_to_flt_6ch_sse4);
- }
- if (EXTERNAL_AVX_FAST(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
- 0, 32, 16, "AVX", ff_conv_s32_to_flt_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
- 0, 32, 32, "AVX", ff_conv_flt_to_s32_avx);
- }
- if (EXTERNAL_AVX(cpu_flags)) {
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- 2, 16, 16, "AVX", ff_conv_s16p_to_s16_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- 6, 16, 8, "AVX", ff_conv_s16p_to_s16_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
- 2, 16, 8, "AVX", ff_conv_s16p_to_flt_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
- 6, 16, 4, "AVX", ff_conv_s16p_to_flt_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP,
- 6, 16, 4, "AVX", ff_conv_fltp_to_s16_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- 6, 16, 4, "AVX", ff_conv_fltp_to_flt_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 2, 16, 8, "AVX", ff_conv_s16_to_s16p_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "AVX", ff_conv_s16_to_s16p_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
- 2, 16, 8, "AVX", ff_conv_s16_to_fltp_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
- 6, 16, 4, "AVX", ff_conv_s16_to_fltp_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
- 2, 16, 8, "AVX", ff_conv_flt_to_s16p_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
- 6, 16, 4, "AVX", ff_conv_flt_to_s16p_6ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
- 2, 16, 4, "AVX", ff_conv_flt_to_fltp_2ch_avx);
- ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
- 6, 16, 4, "AVX", ff_conv_flt_to_fltp_6ch_avx);
- }
-}
diff --git a/libavresample/x86/audio_mix.asm b/libavresample/x86/audio_mix.asm
deleted file mode 100644
index fe27d6a6c9..0000000000
--- a/libavresample/x86/audio_mix.asm
+++ /dev/null
@@ -1,511 +0,0 @@
-;******************************************************************************
-;* x86 optimized channel mixing
-;* Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
-;*
-;* 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 "libavutil/x86/x86util.asm"
-%include "util.asm"
-
-SECTION .text
-
-;-----------------------------------------------------------------------------
-; void ff_mix_2_to_1_fltp_flt(float **src, float **matrix, int len,
-; int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-%macro MIX_2_TO_1_FLTP_FLT 0
-cglobal mix_2_to_1_fltp_flt, 3,4,6, src, matrix, len, src1
- mov src1q, [srcq+gprsize]
- mov srcq, [srcq ]
- sub src1q, srcq
- mov matrixq, [matrixq ]
- VBROADCASTSS m4, [matrixq ]
- VBROADCASTSS m5, [matrixq+4]
- ALIGN 16
-.loop:
- mulps m0, m4, [srcq ]
- mulps m1, m5, [srcq+src1q ]
- mulps m2, m4, [srcq+ mmsize]
- mulps m3, m5, [srcq+src1q+mmsize]
- addps m0, m0, m1
- addps m2, m2, m3
- mova [srcq ], m0
- mova [srcq+mmsize], m2
- add srcq, mmsize*2
- sub lend, mmsize*2/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse
-MIX_2_TO_1_FLTP_FLT
-%if HAVE_AVX_EXTERNAL
-INIT_YMM avx
-MIX_2_TO_1_FLTP_FLT
-%endif
-
-;-----------------------------------------------------------------------------
-; void ff_mix_2_to_1_s16p_flt(int16_t **src, float **matrix, int len,
-; int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-%macro MIX_2_TO_1_S16P_FLT 0
-cglobal mix_2_to_1_s16p_flt, 3,4,6, src, matrix, len, src1
- mov src1q, [srcq+gprsize]
- mov srcq, [srcq]
- sub src1q, srcq
- mov matrixq, [matrixq ]
- VBROADCASTSS m4, [matrixq ]
- VBROADCASTSS m5, [matrixq+4]
- ALIGN 16
-.loop:
- mova m0, [srcq ]
- mova m2, [srcq+src1q]
- S16_TO_S32_SX 0, 1
- S16_TO_S32_SX 2, 3
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- cvtdq2ps m2, m2
- cvtdq2ps m3, m3
- mulps m0, m4
- mulps m1, m4
- mulps m2, m5
- mulps m3, m5
- addps m0, m2
- addps m1, m3
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- packssdw m0, m1
- mova [srcq], m0
- add srcq, mmsize
- sub lend, mmsize/2
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-MIX_2_TO_1_S16P_FLT
-INIT_XMM sse4
-MIX_2_TO_1_S16P_FLT
-
-;-----------------------------------------------------------------------------
-; void ff_mix_2_to_1_s16p_q8(int16_t **src, int16_t **matrix, int len,
-; int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-INIT_XMM sse2
-cglobal mix_2_to_1_s16p_q8, 3,4,6, src, matrix, len, src1
- mov src1q, [srcq+gprsize]
- mov srcq, [srcq]
- sub src1q, srcq
- mov matrixq, [matrixq]
- movd m4, [matrixq]
- movd m5, [matrixq]
- SPLATW m4, m4, 0
- SPLATW m5, m5, 1
- pxor m0, m0
- punpcklwd m4, m0
- punpcklwd m5, m0
- ALIGN 16
-.loop:
- mova m0, [srcq ]
- mova m2, [srcq+src1q]
- punpckhwd m1, m0, m0
- punpcklwd m0, m0
- punpckhwd m3, m2, m2
- punpcklwd m2, m2
- pmaddwd m0, m4
- pmaddwd m1, m4
- pmaddwd m2, m5
- pmaddwd m3, m5
- paddd m0, m2
- paddd m1, m3
- psrad m0, 8
- psrad m1, 8
- packssdw m0, m1
- mova [srcq], m0
- add srcq, mmsize
- sub lend, mmsize/2
- jg .loop
- REP_RET
-
-;-----------------------------------------------------------------------------
-; void ff_mix_1_to_2_fltp_flt(float **src, float **matrix, int len,
-; int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-%macro MIX_1_TO_2_FLTP_FLT 0
-cglobal mix_1_to_2_fltp_flt, 3,5,4, src0, matrix0, len, src1, matrix1
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q]
- sub src1q, src0q
- mov matrix1q, [matrix0q+gprsize]
- mov matrix0q, [matrix0q]
- VBROADCASTSS m2, [matrix0q]
- VBROADCASTSS m3, [matrix1q]
- ALIGN 16
-.loop:
- mova m0, [src0q]
- mulps m1, m0, m3
- mulps m0, m0, m2
- mova [src0q ], m0
- mova [src0q+src1q], m1
- add src0q, mmsize
- sub lend, mmsize/4
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse
-MIX_1_TO_2_FLTP_FLT
-%if HAVE_AVX_EXTERNAL
-INIT_YMM avx
-MIX_1_TO_2_FLTP_FLT
-%endif
-
-;-----------------------------------------------------------------------------
-; void ff_mix_1_to_2_s16p_flt(int16_t **src, float **matrix, int len,
-; int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-%macro MIX_1_TO_2_S16P_FLT 0
-cglobal mix_1_to_2_s16p_flt, 3,5,6, src0, matrix0, len, src1, matrix1
- mov src1q, [src0q+gprsize]
- mov src0q, [src0q]
- sub src1q, src0q
- mov matrix1q, [matrix0q+gprsize]
- mov matrix0q, [matrix0q]
- VBROADCASTSS m4, [matrix0q]
- VBROADCASTSS m5, [matrix1q]
- ALIGN 16
-.loop:
- mova m0, [src0q]
- S16_TO_S32_SX 0, 2
- cvtdq2ps m0, m0
- cvtdq2ps m2, m2
- mulps m1, m0, m5
- mulps m0, m0, m4
- mulps m3, m2, m5
- mulps m2, m2, m4
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- packssdw m0, m2
- packssdw m1, m3
- mova [src0q ], m0
- mova [src0q+src1q], m1
- add src0q, mmsize
- sub lend, mmsize/2
- jg .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-MIX_1_TO_2_S16P_FLT
-INIT_XMM sse4
-MIX_1_TO_2_S16P_FLT
-%if HAVE_AVX_EXTERNAL
-INIT_XMM avx
-MIX_1_TO_2_S16P_FLT
-%endif
-
-;-----------------------------------------------------------------------------
-; void ff_mix_3_8_to_1_2_fltp/s16p_flt(float/int16_t **src, float **matrix,
-; int len, int out_ch, int in_ch);
-;-----------------------------------------------------------------------------
-
-%macro MIX_3_8_TO_1_2_FLT 3 ; %1 = in channels, %2 = out channels, %3 = s16p or fltp
-; define some names to make the code clearer
-%assign in_channels %1
-%assign out_channels %2
-%assign stereo out_channels - 1
-%ifidn %3, s16p
- %assign is_s16 1
-%else
- %assign is_s16 0
-%endif
-
-; determine how many matrix elements must go on the stack vs. mmregs
-%assign matrix_elements in_channels * out_channels
-%if is_s16
- %if stereo
- %assign needed_mmregs 7
- %else
- %assign needed_mmregs 5
- %endif
-%else
- %if stereo
- %assign needed_mmregs 4
- %else
- %assign needed_mmregs 3
- %endif
-%endif
-%assign matrix_elements_mm num_mmregs - needed_mmregs
-%if matrix_elements < matrix_elements_mm
- %assign matrix_elements_mm matrix_elements
-%endif
-%if matrix_elements_mm < matrix_elements
- %assign matrix_elements_stack matrix_elements - matrix_elements_mm
-%else
- %assign matrix_elements_stack 0
-%endif
-%assign matrix_stack_size matrix_elements_stack * mmsize
-
-%assign needed_stack_size -1 * matrix_stack_size
-%if ARCH_X86_32 && in_channels >= 7
-%assign needed_stack_size needed_stack_size - 16
-%endif
-
-cglobal mix_%1_to_%2_%3_flt, 3,in_channels+2,needed_mmregs+matrix_elements_mm, needed_stack_size, src0, src1, len, src2, src3, src4, src5, src6, src7
-
-; define src pointers on stack if needed
-%if matrix_elements_stack > 0 && ARCH_X86_32 && in_channels >= 7
- %define src5m [rsp+matrix_stack_size+0]
- %define src6m [rsp+matrix_stack_size+4]
- %define src7m [rsp+matrix_stack_size+8]
-%endif
-
-; load matrix pointers
-%define matrix0q r1q
-%define matrix1q r3q
-%if stereo
- mov matrix1q, [matrix0q+gprsize]
-%endif
- mov matrix0q, [matrix0q]
-
-; define matrix coeff names
-%assign %%i 0
-%assign %%j needed_mmregs
-%rep in_channels
- %if %%i >= matrix_elements_mm
- CAT_XDEFINE mx_stack_0_, %%i, 1
- CAT_XDEFINE mx_0_, %%i, [rsp+(%%i-matrix_elements_mm)*mmsize]
- %else
- CAT_XDEFINE mx_stack_0_, %%i, 0
- CAT_XDEFINE mx_0_, %%i, m %+ %%j
- %assign %%j %%j+1
- %endif
- %assign %%i %%i+1
-%endrep
-%if stereo
-%assign %%i 0
-%rep in_channels
- %if in_channels + %%i >= matrix_elements_mm
- CAT_XDEFINE mx_stack_1_, %%i, 1
- CAT_XDEFINE mx_1_, %%i, [rsp+(in_channels+%%i-matrix_elements_mm)*mmsize]
- %else
- CAT_XDEFINE mx_stack_1_, %%i, 0
- CAT_XDEFINE mx_1_, %%i, m %+ %%j
- %assign %%j %%j+1
- %endif
- %assign %%i %%i+1
-%endrep
-%endif
-
-; load/splat matrix coeffs
-%assign %%i 0
-%rep in_channels
- %if mx_stack_0_ %+ %%i
- VBROADCASTSS m0, [matrix0q+4*%%i]
- mova mx_0_ %+ %%i, m0
- %else
- VBROADCASTSS mx_0_ %+ %%i, [matrix0q+4*%%i]
- %endif
- %if stereo
- %if mx_stack_1_ %+ %%i
- VBROADCASTSS m0, [matrix1q+4*%%i]
- mova mx_1_ %+ %%i, m0
- %else
- VBROADCASTSS mx_1_ %+ %%i, [matrix1q+4*%%i]
- %endif
- %endif
- %assign %%i %%i+1
-%endrep
-
-; load channel pointers to registers as offsets from the first channel pointer
-%if ARCH_X86_64
- movsxd lenq, r2d
-%endif
- shl lenq, 2-is_s16
-%assign %%i 1
-%rep (in_channels - 1)
- %if ARCH_X86_32 && in_channels >= 7 && %%i >= 5
- mov src5q, [src0q+%%i*gprsize]
- add src5q, lenq
- mov src %+ %%i %+ m, src5q
- %else
- mov src %+ %%i %+ q, [src0q+%%i*gprsize]
- add src %+ %%i %+ q, lenq
- %endif
- %assign %%i %%i+1
-%endrep
- mov src0q, [src0q]
- add src0q, lenq
- neg lenq
-.loop:
-; for x86-32 with 7-8 channels we do not have enough gp registers for all src
-; pointers, so we have to load some of them from the stack each time
-%define copy_src_from_stack ARCH_X86_32 && in_channels >= 7 && %%i >= 5
-%if is_s16
- ; mix with s16p input
- mova m0, [src0q+lenq]
- S16_TO_S32_SX 0, 1
- cvtdq2ps m0, m0
- cvtdq2ps m1, m1
- %if stereo
- mulps m2, m0, mx_1_0
- mulps m3, m1, mx_1_0
- %endif
- mulps m0, m0, mx_0_0
- mulps m1, m1, mx_0_0
-%assign %%i 1
-%rep (in_channels - 1)
- %if copy_src_from_stack
- %define src_ptr src5q
- %else
- %define src_ptr src %+ %%i %+ q
- %endif
- %if stereo
- %if copy_src_from_stack
- mov src_ptr, src %+ %%i %+ m
- %endif
- mova m4, [src_ptr+lenq]
- S16_TO_S32_SX 4, 5
- cvtdq2ps m4, m4
- cvtdq2ps m5, m5
- FMULADD_PS m2, m4, mx_1_ %+ %%i, m2, m6
- FMULADD_PS m3, m5, mx_1_ %+ %%i, m3, m6
- FMULADD_PS m0, m4, mx_0_ %+ %%i, m0, m4
- FMULADD_PS m1, m5, mx_0_ %+ %%i, m1, m5
- %else
- %if copy_src_from_stack
- mov src_ptr, src %+ %%i %+ m
- %endif
- mova m2, [src_ptr+lenq]
- S16_TO_S32_SX 2, 3
- cvtdq2ps m2, m2
- cvtdq2ps m3, m3
- FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m4
- FMULADD_PS m1, m3, mx_0_ %+ %%i, m1, m4
- %endif
- %assign %%i %%i+1
-%endrep
- %if stereo
- cvtps2dq m2, m2
- cvtps2dq m3, m3
- packssdw m2, m3
- mova [src1q+lenq], m2
- %endif
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- packssdw m0, m1
- mova [src0q+lenq], m0
-%else
- ; mix with fltp input
- %if stereo || mx_stack_0_0
- mova m0, [src0q+lenq]
- %endif
- %if stereo
- mulps m1, m0, mx_1_0
- %endif
- %if stereo || mx_stack_0_0
- mulps m0, m0, mx_0_0
- %else
- mulps m0, mx_0_0, [src0q+lenq]
- %endif
-%assign %%i 1
-%rep (in_channels - 1)
- %if copy_src_from_stack
- %define src_ptr src5q
- mov src_ptr, src %+ %%i %+ m
- %else
- %define src_ptr src %+ %%i %+ q
- %endif
- ; avoid extra load for mono if matrix is in a mm register
- %if stereo || mx_stack_0_ %+ %%i
- mova m2, [src_ptr+lenq]
- %endif
- %if stereo
- FMULADD_PS m1, m2, mx_1_ %+ %%i, m1, m3
- %endif
- %if stereo || mx_stack_0_ %+ %%i
- FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m2
- %else
- FMULADD_PS m0, mx_0_ %+ %%i, [src_ptr+lenq], m0, m1
- %endif
- %assign %%i %%i+1
-%endrep
- mova [src0q+lenq], m0
- %if stereo
- mova [src1q+lenq], m1
- %endif
-%endif
-
- add lenq, mmsize
- jl .loop
-; zero ymm high halves
-%if mmsize == 32
- vzeroupper
-%endif
- RET
-%endmacro
-
-%macro MIX_3_8_TO_1_2_FLT_FUNCS 0
-%assign %%i 3
-%rep 6
- INIT_XMM sse
- MIX_3_8_TO_1_2_FLT %%i, 1, fltp
- MIX_3_8_TO_1_2_FLT %%i, 2, fltp
- INIT_XMM sse2
- MIX_3_8_TO_1_2_FLT %%i, 1, s16p
- MIX_3_8_TO_1_2_FLT %%i, 2, s16p
- INIT_XMM sse4
- MIX_3_8_TO_1_2_FLT %%i, 1, s16p
- MIX_3_8_TO_1_2_FLT %%i, 2, s16p
- ; do not use ymm AVX or FMA4 in x86-32 for 6 or more channels due to stack alignment issues
- %if HAVE_AVX_EXTERNAL
- %if ARCH_X86_64 || %%i < 6
- INIT_YMM avx
- %else
- INIT_XMM avx
- %endif
- MIX_3_8_TO_1_2_FLT %%i, 1, fltp
- MIX_3_8_TO_1_2_FLT %%i, 2, fltp
- INIT_XMM avx
- MIX_3_8_TO_1_2_FLT %%i, 1, s16p
- MIX_3_8_TO_1_2_FLT %%i, 2, s16p
- %endif
- %if HAVE_FMA4_EXTERNAL
- %if ARCH_X86_64 || %%i < 6
- INIT_YMM fma4
- %else
- INIT_XMM fma4
- %endif
- MIX_3_8_TO_1_2_FLT %%i, 1, fltp
- MIX_3_8_TO_1_2_FLT %%i, 2, fltp
- INIT_XMM fma4
- MIX_3_8_TO_1_2_FLT %%i, 1, s16p
- MIX_3_8_TO_1_2_FLT %%i, 2, s16p
- %endif
- %assign %%i %%i+1
-%endrep
-%endmacro
-
-MIX_3_8_TO_1_2_FLT_FUNCS
diff --git a/libavresample/x86/audio_mix_init.c b/libavresample/x86/audio_mix_init.c
deleted file mode 100644
index 9b86be2847..0000000000
--- a/libavresample/x86/audio_mix_init.c
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 "config.h"
-#include "libavutil/cpu.h"
-#include "libavutil/x86/cpu.h"
-#include "libavresample/audio_mix.h"
-
-void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len,
- int out_ch, int in_ch);
-void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len,
- int out_ch, int in_ch);
-
-void ff_mix_2_to_1_s16p_flt_sse2(int16_t **src, float **matrix, int len,
- int out_ch, int in_ch);
-void ff_mix_2_to_1_s16p_flt_sse4(int16_t **src, float **matrix, int len,
- int out_ch, int in_ch);
-
-void ff_mix_2_to_1_s16p_q8_sse2(int16_t **src, int16_t **matrix,
- int len, int out_ch, int in_ch);
-
-void ff_mix_1_to_2_fltp_flt_sse(float **src, float **matrix, int len,
- int out_ch, int in_ch);
-void ff_mix_1_to_2_fltp_flt_avx(float **src, float **matrix, int len,
- int out_ch, int in_ch);
-
-void ff_mix_1_to_2_s16p_flt_sse2(int16_t **src, float **matrix, int len,
- int out_ch, int in_ch);
-void ff_mix_1_to_2_s16p_flt_sse4(int16_t **src, float **matrix, int len,
- int out_ch, int in_ch);
-void ff_mix_1_to_2_s16p_flt_avx (int16_t **src, float **matrix, int len,
- int out_ch, int in_ch);
-
-#define DEFINE_MIX_3_8_TO_1_2(chan) \
-void ff_mix_ ## chan ## _to_1_fltp_flt_sse(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_fltp_flt_sse(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_s16p_flt_sse2(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_s16p_flt_sse2(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_s16p_flt_sse4(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_s16p_flt_sse4(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_fltp_flt_avx(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_fltp_flt_avx(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_s16p_flt_avx(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_s16p_flt_avx(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_fltp_flt_fma4(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_fltp_flt_fma4(float **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
- \
-void ff_mix_ ## chan ## _to_1_s16p_flt_fma4(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch); \
-void ff_mix_ ## chan ## _to_2_s16p_flt_fma4(int16_t **src, \
- float **matrix, int len, \
- int out_ch, int in_ch);
-
-DEFINE_MIX_3_8_TO_1_2(3)
-DEFINE_MIX_3_8_TO_1_2(4)
-DEFINE_MIX_3_8_TO_1_2(5)
-DEFINE_MIX_3_8_TO_1_2(6)
-DEFINE_MIX_3_8_TO_1_2(7)
-DEFINE_MIX_3_8_TO_1_2(8)
-
-#define SET_MIX_3_8_TO_1_2(chan) \
- if (EXTERNAL_SSE(cpu_flags)) { \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, 16, 4, "SSE", \
- ff_mix_ ## chan ## _to_1_fltp_flt_sse); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, 16, 4, "SSE", \
- ff_mix_## chan ##_to_2_fltp_flt_sse); \
- } \
- if (EXTERNAL_SSE2(cpu_flags)) { \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, 16, 8, "SSE2", \
- ff_mix_ ## chan ## _to_1_s16p_flt_sse2); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, 16, 8, "SSE2", \
- ff_mix_ ## chan ## _to_2_s16p_flt_sse2); \
- } \
- if (EXTERNAL_SSE4(cpu_flags)) { \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, 16, 8, "SSE4", \
- ff_mix_ ## chan ## _to_1_s16p_flt_sse4); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, 16, 8, "SSE4", \
- ff_mix_ ## chan ## _to_2_s16p_flt_sse4); \
- } \
- if (EXTERNAL_AVX(cpu_flags)) { \
- int ptr_align = 32; \
- int smp_align = 8; \
- if (ARCH_X86_32 || chan >= 6) { \
- ptr_align = 16; \
- smp_align = 4; \
- } \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, ptr_align, smp_align, "AVX", \
- ff_mix_ ## chan ## _to_1_fltp_flt_avx); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, ptr_align, smp_align, "AVX", \
- ff_mix_ ## chan ## _to_2_fltp_flt_avx); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, 16, 8, "AVX", \
- ff_mix_ ## chan ## _to_1_s16p_flt_avx); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, 16, 8, "AVX", \
- ff_mix_ ## chan ## _to_2_s16p_flt_avx); \
- } \
- if (EXTERNAL_FMA4(cpu_flags)) { \
- int ptr_align = 32; \
- int smp_align = 8; \
- if (ARCH_X86_32 || chan >= 6) { \
- ptr_align = 16; \
- smp_align = 4; \
- } \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, ptr_align, smp_align, "FMA4", \
- ff_mix_ ## chan ## _to_1_fltp_flt_fma4); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, ptr_align, smp_align, "FMA4", \
- ff_mix_ ## chan ## _to_2_fltp_flt_fma4); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 1, 16, 8, "FMA4", \
- ff_mix_ ## chan ## _to_1_s16p_flt_fma4); \
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\
- chan, 2, 16, 8, "FMA4", \
- ff_mix_ ## chan ## _to_2_s16p_flt_fma4); \
- }
-
-av_cold void ff_audio_mix_init_x86(AudioMix *am)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (EXTERNAL_SSE(cpu_flags)) {
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse);
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 16, 4, "SSE", ff_mix_1_to_2_fltp_flt_sse);
- }
- if (EXTERNAL_SSE2(cpu_flags)) {
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_flt_sse2);
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
- 2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_q8_sse2);
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 16, 8, "SSE2", ff_mix_1_to_2_s16p_flt_sse2);
- }
- if (EXTERNAL_SSE4(cpu_flags)) {
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 16, 8, "SSE4", ff_mix_2_to_1_s16p_flt_sse4);
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 16, 8, "SSE4", ff_mix_1_to_2_s16p_flt_sse4);
- }
- if (EXTERNAL_AVX_FAST(cpu_flags)) {
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx);
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 32, 8, "AVX", ff_mix_1_to_2_fltp_flt_avx);
- }
- if (EXTERNAL_AVX(cpu_flags)) {
- ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
- 1, 2, 16, 8, "AVX", ff_mix_1_to_2_s16p_flt_avx);
- }
-
- SET_MIX_3_8_TO_1_2(3)
- SET_MIX_3_8_TO_1_2(4)
- SET_MIX_3_8_TO_1_2(5)
- SET_MIX_3_8_TO_1_2(6)
- SET_MIX_3_8_TO_1_2(7)
- SET_MIX_3_8_TO_1_2(8)
-}
diff --git a/libavresample/x86/dither.asm b/libavresample/x86/dither.asm
deleted file mode 100644
index d677c7179a..0000000000
--- a/libavresample/x86/dither.asm
+++ /dev/null
@@ -1,117 +0,0 @@
-;******************************************************************************
-;* x86 optimized dithering format conversion
-;* Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
-;*
-;* 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 "libavutil/x86/x86util.asm"
-
-SECTION_RODATA 32
-
-; 1.0f / (2.0f * INT32_MAX)
-pf_dither_scale: times 8 dd 2.32830643762e-10
-
-pf_s16_scale: times 4 dd 32753.0
-
-SECTION .text
-
-;------------------------------------------------------------------------------
-; void ff_quantize(int16_t *dst, float *src, float *dither, int len);
-;------------------------------------------------------------------------------
-
-INIT_XMM sse2
-cglobal quantize, 4,4,3, dst, src, dither, len
- lea lenq, [2*lend]
- add dstq, lenq
- lea srcq, [srcq+2*lenq]
- lea ditherq, [ditherq+2*lenq]
- neg lenq
- mova m2, [pf_s16_scale]
-.loop:
- mulps m0, m2, [srcq+2*lenq]
- mulps m1, m2, [srcq+2*lenq+mmsize]
- addps m0, [ditherq+2*lenq]
- addps m1, [ditherq+2*lenq+mmsize]
- cvtps2dq m0, m0
- cvtps2dq m1, m1
- packssdw m0, m1
- mova [dstq+lenq], m0
- add lenq, mmsize
- jl .loop
- REP_RET
-
-;------------------------------------------------------------------------------
-; void ff_dither_int_to_float_rectangular(float *dst, int *src, int len)
-;------------------------------------------------------------------------------
-
-%macro DITHER_INT_TO_FLOAT_RECTANGULAR 0
-cglobal dither_int_to_float_rectangular, 3,3,3, dst, src, len
- lea lenq, [4*lend]
- add srcq, lenq
- add dstq, lenq
- neg lenq
- mova m0, [pf_dither_scale]
-.loop:
- cvtdq2ps m1, [srcq+lenq]
- cvtdq2ps m2, [srcq+lenq+mmsize]
- mulps m1, m1, m0
- mulps m2, m2, m0
- mova [dstq+lenq], m1
- mova [dstq+lenq+mmsize], m2
- add lenq, 2*mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-DITHER_INT_TO_FLOAT_RECTANGULAR
-INIT_YMM avx
-DITHER_INT_TO_FLOAT_RECTANGULAR
-
-;------------------------------------------------------------------------------
-; void ff_dither_int_to_float_triangular(float *dst, int *src0, int len)
-;------------------------------------------------------------------------------
-
-%macro DITHER_INT_TO_FLOAT_TRIANGULAR 0
-cglobal dither_int_to_float_triangular, 3,4,5, dst, src0, len, src1
- lea lenq, [4*lend]
- lea src1q, [src0q+2*lenq]
- add src0q, lenq
- add dstq, lenq
- neg lenq
- mova m0, [pf_dither_scale]
-.loop:
- cvtdq2ps m1, [src0q+lenq]
- cvtdq2ps m2, [src0q+lenq+mmsize]
- cvtdq2ps m3, [src1q+lenq]
- cvtdq2ps m4, [src1q+lenq+mmsize]
- addps m1, m1, m3
- addps m2, m2, m4
- mulps m1, m1, m0
- mulps m2, m2, m0
- mova [dstq+lenq], m1
- mova [dstq+lenq+mmsize], m2
- add lenq, 2*mmsize
- jl .loop
- REP_RET
-%endmacro
-
-INIT_XMM sse2
-DITHER_INT_TO_FLOAT_TRIANGULAR
-INIT_YMM avx
-DITHER_INT_TO_FLOAT_TRIANGULAR
diff --git a/libavresample/x86/dither_init.c b/libavresample/x86/dither_init.c
deleted file mode 100644
index ad157b96b1..0000000000
--- a/libavresample/x86/dither_init.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
- *
- * 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 "config.h"
-#include "libavutil/cpu.h"
-#include "libavutil/x86/cpu.h"
-#include "libavresample/dither.h"
-
-void ff_quantize_sse2(int16_t *dst, const float *src, float *dither, int len);
-
-void ff_dither_int_to_float_rectangular_sse2(float *dst, int *src, int len);
-void ff_dither_int_to_float_rectangular_avx(float *dst, int *src, int len);
-
-void ff_dither_int_to_float_triangular_sse2(float *dst, int *src0, int len);
-void ff_dither_int_to_float_triangular_avx(float *dst, int *src0, int len);
-
-av_cold void ff_dither_init_x86(DitherDSPContext *ddsp,
- enum AVResampleDitherMethod method)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (EXTERNAL_SSE2(cpu_flags)) {
- ddsp->quantize = ff_quantize_sse2;
- ddsp->ptr_align = 16;
- ddsp->samples_align = 8;
- }
-
- if (method == AV_RESAMPLE_DITHER_RECTANGULAR) {
- if (EXTERNAL_SSE2(cpu_flags)) {
- ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_sse2;
- }
- if (EXTERNAL_AVX_FAST(cpu_flags)) {
- ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_avx;
- }
- } else {
- if (EXTERNAL_SSE2(cpu_flags)) {
- ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_sse2;
- }
- if (EXTERNAL_AVX_FAST(cpu_flags)) {
- ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_avx;
- }
- }
-}
diff --git a/libavresample/x86/util.asm b/libavresample/x86/util.asm
deleted file mode 100644
index 187a4a21ba..0000000000
--- a/libavresample/x86/util.asm
+++ /dev/null
@@ -1,41 +0,0 @@
-;******************************************************************************
-;* x86 utility macros for libavresample
-;* Copyright (c) 2012 Justin Ruggles <justin.ruggles at gmail.com>
-;*
-;* 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
-;******************************************************************************
-
-%macro S16_TO_S32_SX 2 ; src/low dst, high dst
-%if cpuflag(sse4)
- pmovsxwd m%2, m%1
- psrldq m%1, 8
- pmovsxwd m%1, m%1
- SWAP %1, %2
-%else
- mova m%2, m%1
- punpckhwd m%2, m%2
- punpcklwd m%1, m%1
- psrad m%2, 16
- psrad m%1, 16
-%endif
-%endmacro
-
-%macro DEINT2_PS 3 ; src0/even dst, src1/odd dst, temp
- shufps m%3, m%1, m%2, q3131
- shufps m%1, m%2, q2020
- SWAP %2,%3
-%endmacro
diff --git a/libavresample/x86/w64xmmtest.c b/libavresample/x86/w64xmmtest.c
deleted file mode 100644
index 0f42bd185c..0000000000
--- a/libavresample/x86/w64xmmtest.c
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * check XMM registers for clobbers on Win64
- * Copyright (c) 2013 Martin Storsjo
- *
- * 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 "libavresample/avresample.h"
-#include "libavutil/x86/w64xmmtest.h"
-
-wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
- int out_plane_size, int out_samples, uint8_t **input,
- int in_plane_size, int in_samples))
-{
- testxmmclobbers(avresample_convert, avr, output, out_plane_size,
- out_samples, input, in_plane_size, in_samples);
-}
diff --git a/tests/Makefile b/tests/Makefile
index 7844901e53..d726484b3a 100644
--- a/tests/Makefile
+++ b/tests/Makefile
@@ -157,7 +157,6 @@ include $(SRC_PATH)/tests/fate/indeo.mak
include $(SRC_PATH)/tests/fate/libavcodec.mak
include $(SRC_PATH)/tests/fate/libavdevice.mak
include $(SRC_PATH)/tests/fate/libavformat.mak
-include $(SRC_PATH)/tests/fate/libavresample.mak
include $(SRC_PATH)/tests/fate/libavutil.mak
include $(SRC_PATH)/tests/fate/libswresample.mak
include $(SRC_PATH)/tests/fate/libswscale.mak
diff --git a/tests/fate.sh b/tests/fate.sh
index 0edee7f22e..fc604559cc 100755
--- a/tests/fate.sh
+++ b/tests/fate.sh
@@ -48,7 +48,6 @@ configure()(
--samples="${samples}" \
--enable-gpl \
--enable-memory-poisoning \
- --enable-avresample \
${ignore_tests:+--ignore-tests="$ignore_tests"} \
${arch:+--arch=$arch} \
${cpu:+--cpu="$cpu"} \
diff --git a/tests/fate/libavresample.mak b/tests/fate/libavresample.mak
deleted file mode 100644
index da5cbb35f7..0000000000
--- a/tests/fate/libavresample.mak
+++ /dev/null
@@ -1,68 +0,0 @@
-CROSS_TEST = $(foreach I,$(1), \
- $(foreach J,$(1), \
- $(if $(filter-out $(I),$(J)), \
- $(eval $(call $(2),$(I),$(J),$(3),$(4),$(5))), \
- )))
-
-MIX_CHANNELS = 1 2 3 4 5 6 7 8
-
-define MIX
-FATE_LAVR_MIX += fate-lavr-mix-$(3)-$(1)-$(2)
-fate-lavr-mix-$(3)-$(1)-$(2): tests/data/asynth-44100-$(1).wav
-fate-lavr-mix-$(3)-$(1)-$(2): CMD = ffmpeg -i $(TARGET_PATH)/tests/data/asynth-44100-$(1).wav -ac $(2) -mix_coeff_type $(3) -internal_sample_fmt $(4) -f s16le -af atrim=end_sample=1024 -
-fate-lavr-mix-$(3)-$(1)-$(2): CMP = oneoff
-fate-lavr-mix-$(3)-$(1)-$(2): REF = $(SAMPLES)/lavr/lavr-mix-$(3)-$(1)-$(2)
-endef
-
-$(call CROSS_TEST,$(MIX_CHANNELS),MIX,q8,s16p)
-$(call CROSS_TEST,$(MIX_CHANNELS),MIX,q15,s16p)
-$(call CROSS_TEST,$(MIX_CHANNELS),MIX,flt,fltp)
-
-# test output zeroing with skipped corresponding input
-FATE_LAVR_MIX-$(call FILTERDEMDECENCMUX, CHANNELMAP RESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-lavr-mix-output-zero
-fate-lavr-mix-output-zero: tests/data/filtergraphs/lavr_mix_output_zero tests/data/asynth-44100-4.wav
-fate-lavr-mix-output-zero: CMP = oneoff
-fate-lavr-mix-output-zero: CMD = ffmpeg -i $(TARGET_PATH)/tests/data/asynth-44100-4.wav -filter_script $(TARGET_PATH)/tests/data/filtergraphs/lavr_mix_output_zero -f s16le -
-fate-lavr-mix-output-zero: REF = $(SAMPLES)/lavr/lavr-mix-output-zero
-
-FATE_LAVR_MIX-$(call FILTERDEMDECENCMUX, RESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_LAVR_MIX)
-fate-lavr-mix: $(FATE_LAVR_MIX-yes)
-#FATE_LAVR += $(FATE_LAVR_MIX-yes)
-
-SAMPLERATES = 2626 8000 44100 48000 96000
-
-define RESAMPLE
-FATE_LAVR_RESAMPLE += fate-lavr-resample-$(3)-$(1)-$(2)
-fate-lavr-resample-$(3)-$(1)-$(2): tests/data/asynth-$(1)-1.wav
-fate-lavr-resample-$(3)-$(1)-$(2): CMD = ffmpeg -i $(TARGET_PATH)/tests/data/asynth-$(1)-1.wav -ar $(2) -internal_sample_fmt $(3) -f $(4) -af atrim=end_sample=10240 -
-fate-lavr-resample-$(3)-$(1)-$(2): CMP = oneoff
-fate-lavr-resample-$(3)-$(1)-$(2): CMP_UNIT = $(5)
-fate-lavr-resample-$(3)-$(1)-$(2): FUZZ = 6
-fate-lavr-resample-$(3)-$(1)-$(2): REF = $(SAMPLES)/lavr/lavr-resample-$(3)-$(1)-$(2)-v3
-endef
-
-$(call CROSS_TEST,$(SAMPLERATES),RESAMPLE,s16p,s16le,s16)
-$(call CROSS_TEST,$(SAMPLERATES),RESAMPLE,s32p,s32le,s16)
-$(call CROSS_TEST,$(SAMPLERATES),RESAMPLE,fltp,f32le,f32)
-$(call CROSS_TEST,$(SAMPLERATES),RESAMPLE,dblp,f64le,f64)
-
-FATE_LAVR_RESAMPLE += fate-lavr-resample-linear
-fate-lavr-resample-linear: tests/data/asynth-44100-1.wav
-fate-lavr-resample-linear: CMD = ffmpeg -i $(TARGET_PATH)/tests/data/asynth-44100-1.wav -ar 48000 -filter_size 32 -linear_interp 1 -f s16le -af atrim=end_sample=10240 -
-fate-lavr-resample-linear: CMP = oneoff
-fate-lavr-resample-linear: CMP_UNIT = s16
-fate-lavr-resample-linear: REF = $(SAMPLES)/lavr/lavr-resample-linear
-
-FATE_LAVR_RESAMPLE += fate-lavr-resample-nearest
-fate-lavr-resample-nearest: tests/data/asynth-48000-1.wav
-fate-lavr-resample-nearest: CMD = ffmpeg -i $(TARGET_PATH)/tests/data/asynth-48000-1.wav -ar 44100 -filter_size 0 -phase_shift 0 -f s16le -af atrim=end_sample=10240 -
-fate-lavr-resample-nearest: CMP = oneoff
-fate-lavr-resample-nearest: CMP_UNIT = s16
-fate-lavr-resample-nearest: REF = $(SAMPLES)/lavr/lavr-resample-nearest
-
-FATE_LAVR_RESAMPLE-$(call FILTERDEMDECENCMUX, RESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_LAVR_RESAMPLE)
-fate-lavr-resample: $(FATE_LAVR_RESAMPLE-yes)
-#FATE_LAVR += $(FATE_LAVR_RESAMPLE-yes)
-
-FATE_SAMPLES_AVCONV += $(FATE_LAVR)
-fate-lavr: $(FATE_LAVR)
diff --git a/tools/gen-rc b/tools/gen-rc
index d9ca37e9ff..a28b013aae 100755
--- a/tools/gen-rc
+++ b/tools/gen-rc
@@ -43,7 +43,6 @@ EOF
# gen-rc libavdevice "FFmpeg device handling library"
# gen-rc libavfilter "FFmpeg audio/video filtering library"
# gen-rc libpostproc "FFmpeg postprocessing library"
-# gen-rc libavresample "Libav audio resampling library"
# gen-rc libswscale "FFmpeg image rescaling library"
# gen-rc libswresample "FFmpeg audio resampling library"
diff --git a/tools/target_dec_fuzzer.c b/tools/target_dec_fuzzer.c
index 80da7afcb8..334c47a2c8 100644
--- a/tools/target_dec_fuzzer.c
+++ b/tools/target_dec_fuzzer.c
@@ -30,7 +30,7 @@
* build the fuzz target.
Choose the value of FFMPEG_CODEC (e.g. AV_CODEC_ID_DVD_SUBTITLE) and
choose one of FUZZ_FFMPEG_VIDEO, FUZZ_FFMPEG_AUDIO, FUZZ_FFMPEG_SUBTITLE.
- clang -fsanitize=address -fsanitize-coverage=trace-pc-guard,trace-cmp tools/target_dec_fuzzer.c -o target_dec_fuzzer -I. -DFFMPEG_CODEC=AV_CODEC_ID_MPEG1VIDEO -DFUZZ_FFMPEG_VIDEO ../../libfuzzer/libFuzzer.a -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,-z,noexecstack -Wl,--warn-common -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -ldl -lxcb -lxcb-shm -lxcb -lxcb-xfixes -lxcb -lxcb-shape -lxcb -lX11 -lasound -lm -lbz2 -lz -pthread
+ clang -fsanitize=address -fsanitize-coverage=trace-pc-guard,trace-cmp tools/target_dec_fuzzer.c -o target_dec_fuzzer -I. -DFFMPEG_CODEC=AV_CODEC_ID_MPEG1VIDEO -DFUZZ_FFMPEG_VIDEO ../../libfuzzer/libFuzzer.a -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,-z,noexecstack -Wl,--warn-common -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -ldl -lxcb -lxcb-shm -lxcb -lxcb-xfixes -lxcb -lxcb-shape -lxcb -lX11 -lasound -lm -lbz2 -lz -pthread
* create a corpus directory and put some samples there (empty dir is ok too):
mkdir CORPUS && cp some-files CORPUS
--
2.31.1
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