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-rwxr-xr-xconfigure1
-rw-r--r--doc/filters.texi4
-rw-r--r--libavcodec/x86/vc1dsp_mmx.c2
-rw-r--r--libavcodec/x86/vc1dsp_yasm.asm11
-rw-r--r--libavfilter/Makefile2
-rw-r--r--libavfilter/af_resample.c225
-rw-r--r--libavfilter/allfilters.c1
7 files changed, 233 insertions, 13 deletions
diff --git a/configure b/configure
index 270de65f28..783c9e0775 100755
--- a/configure
+++ b/configure
@@ -1690,6 +1690,7 @@ movie_filter_deps="avcodec avformat"
mp_filter_deps="gpl avcodec swscale postproc"
mptestsrc_filter_deps="gpl"
negate_filter_deps="lut_filter"
+resample_filter_deps="avresample"
ocv_filter_deps="libopencv"
pan_filter_deps="swresample"
removelogo_filter_deps="avcodec avformat swscale"
diff --git a/doc/filters.texi b/doc/filters.texi
index ef65d101a7..2af7b37d07 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -502,6 +502,10 @@ volume=-12dB
@end example
@end itemize
+@section resample
+Convert the audio sample format, sample rate and channel layout. This filter is
+not meant to be used directly.
+
@c man end AUDIO FILTERS
@chapter Audio Sources
diff --git a/libavcodec/x86/vc1dsp_mmx.c b/libavcodec/x86/vc1dsp_mmx.c
index 32891a02fe..d1cb852f09 100644
--- a/libavcodec/x86/vc1dsp_mmx.c
+++ b/libavcodec/x86/vc1dsp_mmx.c
@@ -701,7 +701,6 @@ static void vc1_h_loop_filter16_ ## EXT(uint8_t *src, int stride, int pq) \
}
#if HAVE_YASM
-LOOP_FILTER(mmx)
LOOP_FILTER(mmx2)
LOOP_FILTER(sse2)
LOOP_FILTER(ssse3)
@@ -803,7 +802,6 @@ void ff_vc1dsp_init_mmx(VC1DSPContext *dsp)
#if HAVE_YASM
if (mm_flags & AV_CPU_FLAG_MMX) {
- ASSIGN_LF(mmx);
}
return;
if (mm_flags & AV_CPU_FLAG_MMX2) {
diff --git a/libavcodec/x86/vc1dsp_yasm.asm b/libavcodec/x86/vc1dsp_yasm.asm
index 1eba3c1198..b897580b76 100644
--- a/libavcodec/x86/vc1dsp_yasm.asm
+++ b/libavcodec/x86/vc1dsp_yasm.asm
@@ -227,13 +227,6 @@ section .text
imul r2, 0x01010101
%endmacro
-; I do not know why the sign extension is needed...
-%macro PSIGNW_SRA_MMX 2
- psraw %2, 15
- PSIGNW_MMX %1, %2
-%endmacro
-
-
%macro VC1_LF_MMX 1
INIT_MMX
cglobal vc1_v_loop_filter_internal_%1
@@ -274,10 +267,6 @@ cglobal vc1_h_loop_filter8_%1, 3,5,0
RET
%endmacro
-%define PABSW PABSW_MMX
-%define PSIGNW PSIGNW_SRA_MMX
-VC1_LF_MMX mmx
-
%define PABSW PABSW_MMX2
VC1_LF_MMX mmx2
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 962dbf63a9..70f2c9e5ca 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -2,6 +2,7 @@ include $(SUBDIR)../config.mak
NAME = avfilter
FFLIBS = avutil swscale
+FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
@@ -48,6 +49,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER) += af_asplit.o
OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
+OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
new file mode 100644
index 0000000000..f46e24b1b6
--- /dev/null
+++ b/libavfilter/af_resample.c
@@ -0,0 +1,225 @@
+/*
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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/mathematics.h"
+#include "libavutil/opt.h"
+
+#include "libavresample/avresample.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct ResampleContext {
+ AVAudioResampleContext *avr;
+
+ int64_t next_pts;
+} ResampleContext;
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ResampleContext *s = ctx->priv;
+
+ if (s->avr) {
+ avresample_close(s->avr);
+ avresample_free(&s->avr);
+ }
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+
+ AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
+ AVFilterFormats *out_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
+
+ avfilter_formats_ref(in_formats, &inlink->out_formats);
+ avfilter_formats_ref(out_formats, &outlink->in_formats);
+
+ 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;
+
+ 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)
+ return 0;
+
+ if (!(s->avr = avresample_alloc_context()))
+ return AVERROR(ENOMEM);
+
+ 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 both the input and output formats are s16 or u8, use s16 as
+ the internal sample format */
+ if (av_get_bytes_per_sample(inlink->format) <= 2 &&
+ av_get_bytes_per_sample(outlink->format) <= 2)
+ av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
+
+ if ((ret = avresample_open(s->avr)) < 0)
+ return ret;
+
+ outlink->time_base = (AVRational){ 1, outlink->sample_rate };
+ s->next_pts = AV_NOPTS_VALUE;
+
+ 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 = avfilter_request_frame(ctx->inputs[0]);
+
+ /* flush the lavr delay buffer */
+ if (ret == AVERROR_EOF && s->avr) {
+ AVFilterBufferRef *buf;
+ int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
+ outlink->sample_rate,
+ ctx->inputs[0]->sample_rate,
+ AV_ROUND_UP);
+
+ if (!nb_samples)
+ return ret;
+
+ buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ if (!buf)
+ return AVERROR(ENOMEM);
+
+ ret = avresample_convert(s->avr, (void**)buf->extended_data,
+ buf->linesize[0], nb_samples,
+ NULL, 0, 0);
+ if (ret <= 0) {
+ avfilter_unref_buffer(buf);
+ return (ret == 0) ? AVERROR_EOF : ret;
+ }
+
+ buf->pts = s->next_pts;
+ ff_filter_samples(outlink, buf);
+ return 0;
+ }
+ return ret;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ResampleContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+
+ if (s->avr) {
+ AVFilterBufferRef *buf_out;
+ int delay, nb_samples, ret;
+
+ /* maximum possible samples lavr can output */
+ delay = avresample_get_delay(s->avr);
+ nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
+ outlink->sample_rate, inlink->sample_rate,
+ AV_ROUND_UP);
+
+ buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
+ buf_out->linesize[0], nb_samples,
+ (void**)buf->extended_data, buf->linesize[0],
+ buf->audio->nb_samples);
+
+ av_assert0(!avresample_available(s->avr));
+
+ if (s->next_pts == AV_NOPTS_VALUE) {
+ if (buf->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(buf->pts, inlink->time_base,
+ outlink->time_base);
+ }
+
+ if (ret > 0) {
+ buf_out->audio->nb_samples = ret;
+ if (buf->pts != AV_NOPTS_VALUE) {
+ buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
+ outlink->time_base) -
+ av_rescale(delay, outlink->sample_rate,
+ inlink->sample_rate);
+ } else
+ buf_out->pts = s->next_pts;
+
+ s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
+
+ ff_filter_samples(outlink, buf_out);
+ }
+ avfilter_unref_buffer(buf);
+ } else
+ ff_filter_samples(outlink, buf);
+}
+
+AVFilter avfilter_af_resample = {
+ .name = "resample",
+ .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
+ .priv_size = sizeof(ResampleContext),
+
+ .uninit = uninit,
+ .query_formats = query_formats,
+
+ .inputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ },
+ { .name = NULL}},
+ .outputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame },
+ { .name = NULL}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index e4f82c96c5..4e4c5d37f4 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -46,6 +46,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (PAN, pan, af);
REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
REGISTER_FILTER (VOLUME, volume, af);
+ REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-22 10:35:01 +0000