[FFmpeg-cvslog] lavfi: add asetnsamples audio filter
Stefano Sabatini
git at videolan.org
Sat Jun 16 00:57:03 CEST 2012
ffmpeg | branch: master | Stefano Sabatini <stefasab at gmail.com> | Fri May 25 13:14:53 2012 +0200| [2b1fc5621d93b04d6b78b69402c7a796ceac35ed] | committer: Stefano Sabatini
lavfi: add asetnsamples audio filter
This filter changes the number of samples on single output operation.
Based on a patch by Andrey Utkin <andrey.krieger.utkin at gmail.com>.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2b1fc5621d93b04d6b78b69402c7a796ceac35ed
---
Changelog | 1 +
doc/filters.texi | 30 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_asetnsamples.c | 206 +++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 240 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 026d102..facd0e9 100644
--- a/Changelog
+++ b/Changelog
@@ -6,6 +6,7 @@ version next:
- Scene detection in libavfilter
- Indeo Audio decoder
- channelsplit audio filter
+- setnsamples audio filter
version 0.11:
diff --git a/doc/filters.texi b/doc/filters.texi
index 67242af..f2767cb 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -273,6 +273,36 @@ For example, to resample the input audio to 44100Hz:
aresample=44100
@end example
+ at section asetnsamples
+
+Set the number of samples per each output audio frame.
+
+The last output packet may contain a different number of samples, as
+the filter will flush all the remaining samples when the input audio
+signal its end.
+
+The filter accepts parameters as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
+ at table @option
+
+ at item nb_out_samples, n
+Set the number of frames per each output audio frame. The number is
+intended as the number of samples @emph{per each channel}.
+Default value is 1024.
+
+ at item pad, p
+If set to 1, the filter will pad the last audio frame with zeroes, so
+that the last frame will contain the same number of samples as the
+previous ones. Default value is 1.
+ at end table
+
+For example, to set the number of per-frame samples to 1234 and
+disable padding for the last frame, use:
+ at example
+asetnsamples=n=1234:p=0
+ at end example
+
@section ashowinfo
Show a line containing various information for each input audio frame.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 95126a8..fee7641 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -51,6 +51,7 @@ OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
+OBJS-$(CONFIG_ASETNSAMPLES_FILTER) += af_asetnsamples.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
diff --git a/libavfilter/af_asetnsamples.c b/libavfilter/af_asetnsamples.c
new file mode 100644
index 0000000..eb3c6a9
--- /dev/null
+++ b/libavfilter/af_asetnsamples.c
@@ -0,0 +1,206 @@
+/*
+ * Copyright (c) 2012 Andrey Utkin
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * 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
+ * Filter that changes number of samples on single output operation
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct {
+ const AVClass *class;
+ int nb_out_samples; ///< how many samples to output
+ AVAudioFifo *fifo; ///< samples are queued here
+ int64_t next_out_pts;
+ int req_fullfilled;
+ int pad;
+} ASNSContext;
+
+#define OFFSET(x) offsetof(ASNSContext, x)
+
+static const AVOption asns_options[] = {
+{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+{ "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
+{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
+{ NULL }
+};
+
+static const AVClass asns_class = {
+ "asetnsamples",
+ av_default_item_name,
+ asns_options
+};
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ ASNSContext *asns = ctx->priv;
+ int err;
+
+ asns->class = &asns_class;
+ av_opt_set_defaults(asns);
+
+ if ((err = av_set_options_string(asns, args, "=", ":")) < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
+ return err;
+ }
+
+ asns->next_out_pts = AV_NOPTS_VALUE;
+ av_log(ctx, AV_LOG_INFO, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ASNSContext *asns = ctx->priv;
+ av_audio_fifo_free(asns->fifo);
+}
+
+static int config_props_output(AVFilterLink *outlink)
+{
+ ASNSContext *asns = outlink->src->priv;
+ int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+
+ asns->fifo = av_audio_fifo_alloc(outlink->format, nb_channels, asns->nb_out_samples);
+ if (!asns->fifo)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static int push_samples(AVFilterLink *outlink)
+{
+ ASNSContext *asns = outlink->src->priv;
+ AVFilterBufferRef *outsamples = NULL;
+ int nb_out_samples, nb_pad_samples;
+
+ if (asns->pad) {
+ nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
+ nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
+ } else {
+ nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
+ nb_pad_samples = 0;
+ }
+
+ if (!nb_out_samples)
+ return 0;
+
+ outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_out_samples);
+ av_assert0(outsamples);
+
+ av_audio_fifo_read(asns->fifo,
+ (void **)outsamples->extended_data, nb_out_samples);
+
+ if (nb_pad_samples)
+ av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
+ nb_pad_samples, av_get_channel_layout_nb_channels(outlink->channel_layout),
+ outlink->format);
+ outsamples->audio->nb_samples = nb_out_samples;
+ outsamples->audio->channel_layout = outlink->channel_layout;
+ outsamples->audio->sample_rate = outlink->sample_rate;
+ outsamples->pts = asns->next_out_pts;
+
+ if (asns->next_out_pts != AV_NOPTS_VALUE)
+ asns->next_out_pts += nb_out_samples;
+
+ ff_filter_samples(outlink, outsamples);
+ asns->req_fullfilled = 1;
+ return nb_out_samples;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ASNSContext *asns = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int ret;
+ int nb_samples = insamples->audio->nb_samples;
+
+ if (av_audio_fifo_space(asns->fifo) < nb_samples) {
+ av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
+ ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
+ if (ret < 0) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Stretching audio fifo failed, discarded %d samples\n", nb_samples);
+ return;
+ }
+ }
+ av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
+ if (asns->next_out_pts == AV_NOPTS_VALUE)
+ asns->next_out_pts = insamples->pts;
+ avfilter_unref_buffer(insamples);
+
+ if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
+ push_samples(outlink);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ ASNSContext *asns = outlink->src->priv;
+ AVFilterLink *inlink = outlink->src->inputs[0];
+ int ret;
+
+ asns->req_fullfilled = 0;
+ do {
+ ret = avfilter_request_frame(inlink);
+ } while (!asns->req_fullfilled && ret >= 0);
+
+ if (ret == AVERROR_EOF)
+ while (push_samples(outlink))
+ ;
+
+ return ret;
+}
+
+AVFilter avfilter_af_asetnsamples = {
+ .name = "asetnsamples",
+ .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
+ .priv_size = sizeof(ASNSContext),
+ .init = init,
+ .uninit = uninit,
+
+ .inputs = (const AVFilterPad[]) {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ|AV_PERM_WRITE
+ },
+ { .name = NULL }
+ },
+
+ .outputs = (const AVFilterPad[]) {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_props_output,
+ },
+ { .name = NULL }
+ },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f8d6b38..8257ee5 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -40,6 +40,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (AMIX, amix, af);
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
+ REGISTER_FILTER (ASETNSAMPLES, asetnsamples, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (ASPLIT, asplit, af);
REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 3ebea25..861fe99 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 2
-#define LIBAVFILTER_VERSION_MINOR 79
+#define LIBAVFILTER_VERSION_MINOR 80
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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