lavfi: add channelsplit audio filter.

8 files changed, 192 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 898a2473e8..b80ff8885e 100644
--- a/Changelog
+++ b/Changelog
@@ -24,6 +24,7 @@ version <next>:
 - avprobe output is now standard INI or JSON. The old format can still
   be used with -of old.
 - Indeo Audio decoder
+- channelsplit audio filter
 
 
 version 0.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 0179682918..f17dad8051 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -207,6 +207,31 @@ Maximum compensation in samples per second.
 
 @end table
 
+@section channelsplit
+Split each channel in input audio stream into a separate output stream.
+
+This filter accepts the following named parameters:
+@table @option
+@item channel_layout
+Channel layout of the input stream. Default is "stereo".
+@end table
+
+For example, assuming a stereo input MP3 file
+@example
+avconv -i in.mp3 -filter_complex channelsplit out.mkv
+@end example
+will create an output Matroska file with two audio streams, one containing only
+the left channel and the other the right channel.
+
+To split a 5.1 WAV file into per-channel files
+@example
+avconv -i in.wav -filter_complex
+'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
+-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
+front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
+side_right.wav
+@end example
+
 @section resample
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly, it is inserted automatically by libavfilter
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 914f0c6b8e..7e1a6d1209 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -29,6 +29,7 @@ OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(CONFIG_ASYNCTS_FILTER)                += af_asyncts.o
+OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 
 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o
diff --git a/libavfilter/af_channelsplit.c b/libavfilter/af_channelsplit.c
new file mode 100644
index 0000000000..c3e7ccfc69
--- /dev/null
+++ b/libavfilter/af_channelsplit.c
@@ -0,0 +1,146 @@
+/*
+ * 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
+ * Channel split filter
+ *
+ * Split an audio stream into per-channel streams.
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+typedef struct ChannelSplitContext {
+    const AVClass *class;
+
+    uint64_t channel_layout;
+    char    *channel_layout_str;
+} ChannelSplitContext;
+
+#define OFFSET(x) offsetof(ChannelSplitContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+    { "channel_layout", "Input channel layout.", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, { .str = "stereo" }, .flags = A },
+    { NULL },
+};
+
+static const AVClass channelsplit_class = {
+    .class_name = "channelsplit filter",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+static int init(AVFilterContext *ctx, const char *arg, void *opaque)
+{
+    ChannelSplitContext *s = ctx->priv;
+    int nb_channels;
+    int ret = 0, i;
+
+    s->class = &channelsplit_class;
+    av_opt_set_defaults(s);
+    if ((ret = av_set_options_string(s, arg, "=", ":")) < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", arg);
+        return ret;
+    }
+    if (!(s->channel_layout = av_get_channel_layout(s->channel_layout_str))) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing channel layout '%s'.\n",
+               s->channel_layout_str);
+        ret = AVERROR(EINVAL);
+        goto fail;
+    }
+
+    nb_channels = av_get_channel_layout_nb_channels(s->channel_layout);
+    for (i = 0; i < nb_channels; i++) {
+        uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
+        AVFilterPad pad  = { 0 };
+
+        pad.type = AVMEDIA_TYPE_AUDIO;
+        pad.name = av_get_channel_name(channel);
+
+        ff_insert_outpad(ctx, i, &pad);
+    }
+
+fail:
+    av_opt_free(s);
+    return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    ChannelSplitContext *s = ctx->priv;
+    AVFilterChannelLayouts *in_layouts = NULL;
+    int i;
+
+    ff_set_common_formats    (ctx, ff_planar_sample_fmts());
+    ff_set_common_samplerates(ctx, ff_all_samplerates());
+
+    ff_add_channel_layout(&in_layouts, s->channel_layout);
+    ff_channel_layouts_ref(in_layouts, &ctx->inputs[0]->out_channel_layouts);
+
+    for (i = 0; i < ctx->output_count; i++) {
+        AVFilterChannelLayouts *out_layouts = NULL;
+        uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
+
+        ff_add_channel_layout(&out_layouts, channel);
+        ff_channel_layouts_ref(out_layouts, &ctx->outputs[i]->in_channel_layouts);
+    }
+
+    return 0;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    AVFilterContext *ctx = inlink->dst;
+    int i;
+
+    for (i = 0; i < ctx->output_count; i++) {
+        AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
+
+        if (!buf_out)
+            return;
+
+        buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
+        buf_out->audio->channel_layout =
+            av_channel_layout_extract_channel(buf->audio->channel_layout, i);
+
+        ff_filter_samples(ctx->outputs[i], buf_out);
+    }
+    avfilter_unref_buffer(buf);
+}
+
+AVFilter avfilter_af_channelsplit = {
+    .name           = "channelsplit",
+    .description    = NULL_IF_CONFIG_SMALL("Split audio into per-channel streams"),
+    .priv_size      = sizeof(ChannelSplitContext),
+
+    .init           = init,
+    .query_formats  = query_formats,
+
+    .inputs  = (const AVFilterPad[]){{ .name           = "default",
+                                       .type           = AVMEDIA_TYPE_AUDIO,
+                                       .filter_samples = filter_samples, },
+                                     { NULL }},
+    .outputs = (const AVFilterPad[]){{ NULL }},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 941ca6a20d..681487118b 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -39,6 +39,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ASPLIT,      asplit,      af);
     REGISTER_FILTER (ASYNCTS,     asyncts,     af);
+    REGISTER_FILTER (CHANNELSPLIT,channelsplit,af);
     REGISTER_FILTER (RESAMPLE,    resample,    af);
 
     REGISTER_FILTER (ANULLSRC,    anullsrc,    asrc);
diff --git a/libavfilter/formats.c b/libavfilter/formats.c
index 83bdcd7a7a..1d58d95b8e 100644
--- a/libavfilter/formats.c
+++ b/libavfilter/formats.c
@@ -220,6 +220,18 @@ AVFilterFormats *ff_all_formats(enum AVMediaType type)
     return ret;
 }
 
+AVFilterFormats *ff_planar_sample_fmts(void)
+{
+    AVFilterFormats *ret = NULL;
+    int fmt;
+
+    for (fmt = 0; fmt < AV_SAMPLE_FMT_NB; fmt++)
+        if (av_sample_fmt_is_planar(fmt))
+            ff_add_format(&ret, fmt);
+
+    return ret;
+}
+
 AVFilterFormats *ff_all_samplerates(void)
 {
     AVFilterFormats *ret = av_mallocz(sizeof(*ret));
diff --git a/libavfilter/formats.h b/libavfilter/formats.h
index 8961cb1a65..c3e9522c89 100644
--- a/libavfilter/formats.h
+++ b/libavfilter/formats.h
@@ -160,6 +160,11 @@ int ff_add_format(AVFilterFormats **avff, int fmt);
 AVFilterFormats *ff_all_formats(enum AVMediaType type);
 
 /**
+ * Construct a formats list containing all planar sample formats.
+ */
+AVFilterFormats *ff_planar_sample_fmts(void);
+
+/**
  * Return a format list which contains the intersection of the formats of
  * a and b. Also, all the references of a, all the references of b, and
  * a and b themselves will be deallocated.
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 9a64316949..09cf4c69e5 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  20
+#define LIBAVFILTER_VERSION_MINOR  21
 #define LIBAVFILTER_VERSION_MICRO  0
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \