[FFmpeg-devel] [PATCH 2/2] lavfi: add amerge audio filter.

Nicolas George nicolas.george at normalesup.org
Fri Dec 23 14:17:14 CET 2011


Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
---
 Changelog                |    1 +
 doc/filters.texi         |   30 +++++
 libavfilter/Makefile     |    1 +
 libavfilter/af_amerge.c  |  271 ++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |    1 +
 5 files changed, 304 insertions(+), 0 deletions(-)
 create mode 100644 libavfilter/af_amerge.c

diff --git a/Changelog b/Changelog
index 7044b4c..430048e 100644
--- a/Changelog
+++ b/Changelog
@@ -8,6 +8,7 @@ version next:
 - OpenMG Audio muxer
 - SMJPEG demuxer
 - astreamsync audio filter
+- amerge audio filter
 
 
 version 0.9:
diff --git a/doc/filters.texi b/doc/filters.texi
index ce4a74f..77e9c50 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -156,6 +156,36 @@ aformat=u8\\,s16:mono:packed
 aformat=s16:mono\\,stereo:all
 @end example
 
+ at section amerge
+
+Merge two audio streams into a single multi-channel stream.
+
+This filter does not need any argument.
+
+If the channel layouts of the inputs are disjoint, and therefore compatible,
+the channel layout of the output will be set accordingly and the channels
+will be reordered as necessary. If the channel layouts of the inputs are not
+disjoint, the output will have all the channels of the first input then all
+the channels of the second input, in that order, and the channel layout of
+the output will be the default value corresponding to the total number of
+channels.
+
+For example, if the first input is in 2.1 (FL+FR+LF) and the second input
+is FC+BL+BR, then the output will be in 5.1, with the channels in the
+following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
+first input, b1 is the first channel of the second input).
+
+On the other hand, if both input are in stereo, the output channels will be
+in the default order: a1, a2, b1, b2, and the channel layout will be
+arbitrarily set to 4.0, which may or may not be the expected value.
+
+Both inputs must have the same sample rate, format and packing.
+
+Example: merge two mono files into a stereo stream:
+ at example
+amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
+ at end example
+
 @section anull
 
 Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 555767b..445daa1 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -26,6 +26,7 @@ OBJS-$(CONFIG_AVCODEC)                       += avcodec.o
 
 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
+OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c
new file mode 100644
index 0000000..70f82a9
--- /dev/null
+++ b/libavfilter/af_amerge.c
@@ -0,0 +1,271 @@
+/*
+ * Copyright (c) 2011 Nicolas George <nicolas.george at normalesup.org>
+ *
+ * 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio merging filter
+ */
+
+#include <stdlib.h>
+#include "libavcodec/avcodec.h"
+#include "libavutil/avstring.h"
+#include "libswresample/swresample.h" // only for SWR_CH_MAX
+#include "avfilter.h"
+#include "internal.h"
+
+#define QUEUE_SIZE 16
+
+struct amerge_context {
+    int route[SWR_CH_MAX];
+    int nb_in_ch[2];
+    int bps;
+    struct amerge_queue {
+        AVFilterBufferRef *buf[QUEUE_SIZE];
+        int nb_buf, nb_samples, pos;
+    } queue[2];
+};
+
+static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
+{
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    struct amerge_context *am = ctx->priv;
+    int i, j;
+
+    for (i = 0; i < 2; i++)
+        for (j = 0; j < am->queue[i].nb_buf; j++)
+            avfilter_unref_buffer(am->queue[i].buf[j]);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    struct amerge_context *am = ctx->priv;
+    int64_t inlayout[2], outlayout;
+    const int packing_fmts[] = { AVFILTER_PACKED, -1 };
+    AVFilterFormats *formats;
+    int i;
+
+    for (i = 0; i < 2; i++) {
+        if (!ctx->inputs[i]->in_chlayouts ||
+            !ctx->inputs[i]->in_chlayouts->format_count) {
+            av_log(ctx, AV_LOG_ERROR,
+                   "No channel layout for input %d\n", i + 1);
+            return AVERROR(EINVAL);
+        }
+        inlayout[i] = ctx->inputs[i]->in_chlayouts->formats[0];
+        if (ctx->inputs[i]->in_chlayouts->format_count > 1) {
+            char buf[256];
+            av_get_channel_layout_string(buf, sizeof(buf), 0, inlayout[i]);
+            av_log(ctx, AV_LOG_INFO, "Using \"%s\" for input %d\n", buf, i + 1);
+        }
+        am->nb_in_ch[i] = av_get_channel_layout_nb_channels(inlayout[i]);
+    }
+    if (am->nb_in_ch[0] + am->nb_in_ch[1] > SWR_CH_MAX) {
+        av_log(ctx, AV_LOG_ERROR, "Too many channels\n");
+        return AVERROR(EINVAL);
+    }
+    if (inlayout[0] & inlayout[1]) {
+        av_log(ctx, AV_LOG_WARNING,
+               "Inputs overlap: output layout will be meaningless\n");
+        for (i = 0; i < am->nb_in_ch[0] + am->nb_in_ch[1]; i++)
+            am->route[i] = i;
+        outlayout = av_get_default_channel_layout(am->nb_in_ch[0] +
+                                                  am->nb_in_ch[1]);
+        if (!outlayout)
+            outlayout = ((int64_t)1 << (am->nb_in_ch[0] + am->nb_in_ch[1])) - 1;
+    } else {
+        int *route[2] = { am->route, am->route + am->nb_in_ch[0] };
+        int c, out_no = 0;
+
+        outlayout = inlayout[0] | inlayout[1];
+        for (c = 0; c < 64; c++)
+            for (i = 0; i < 2; i++)
+                if ((inlayout[i] >> c) & 1)
+                    *(route[i]++) = out_no++;
+    }
+    formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
+    avfilter_set_common_sample_formats(ctx, formats);
+    formats = avfilter_make_format_list(packing_fmts);
+    avfilter_set_common_packing_formats(ctx, formats);
+    for (i = 0; i < 2; i++) {
+        formats = NULL;
+        avfilter_add_format(&formats, inlayout[i]);
+        avfilter_formats_ref(formats, &ctx->inputs[i]->out_chlayouts);
+    }
+    formats = NULL;
+    avfilter_add_format(&formats, outlayout);
+    avfilter_formats_ref(formats, &ctx->outputs[0]->in_chlayouts);
+    return 0;
+}
+
+static int config_props(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    struct amerge_context *am = ctx->priv;
+
+    if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Inputs must have the same sample rate "
+               "(%"PRIi64" vs %"PRIi64")\n",
+               ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
+        return AVERROR(EINVAL);
+    }
+    am->bps = av_get_bytes_per_sample(ctx->outputs[0]->format);
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    struct amerge_context *am = ctx->priv;
+    int i;
+
+    for (i = 0; i < 2; i++)
+        if (!am->queue[i].nb_samples)
+            avfilter_request_frame(ctx->inputs[i]);
+    return 0;
+}
+
+static inline void copy_samples(int nb_in_ch[2], int *route0, uint8_t *ins[2],
+                                uint8_t **outs, int ns, int bps)
+{
+    int *route;
+    int i, c;
+
+    while (ns--) {
+        route = route0;
+        for (i = 0; i < 2; i++) {
+            for (c = 0; c < nb_in_ch[i]; c++) {
+                memcpy((*outs) + bps * *(route++), ins[i], bps);
+                ins[i] += bps;
+            }
+        }
+        *outs += (nb_in_ch[0] + nb_in_ch[1]) * bps;
+    }
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+    AVFilterContext *ctx = inlink->dst;
+    struct amerge_context *am = ctx->priv;
+    int in_no = inlink == ctx->inputs[1];
+    struct amerge_queue *iq = &am->queue[in_no];
+    int nb_samples, ns, i;
+    AVFilterBufferRef *outbuf, **ibuf[2];
+    uint8_t *ins[2], *outs;
+
+    if (iq->nb_buf == QUEUE_SIZE) {
+        av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n");
+        avfilter_unref_buffer(insamples);
+        return;
+    }
+    iq->buf[iq->nb_buf++] = avfilter_ref_buffer(insamples, AV_PERM_READ |
+                                                           AV_PERM_PRESERVE);
+    iq->nb_samples += insamples->audio->nb_samples;
+    avfilter_unref_buffer(insamples);
+    if (!am->queue[!in_no].nb_samples)
+        return;
+
+    nb_samples = FFMIN(am->queue[0].nb_samples,
+                       am->queue[1].nb_samples);
+    outbuf = avfilter_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE,
+                                       nb_samples);
+    outs = outbuf->data[0];
+    for (i = 0; i < 2; i++) {
+        ibuf[i] = am->queue[i].buf;
+        ins[i] = (*ibuf[i])->data[0] +
+                 am->queue[i].pos * am->nb_in_ch[i] * am->bps;
+    }
+    while (nb_samples) {
+        ns = nb_samples;
+        for (i = 0; i < 2; i++)
+            ns = FFMIN(ns, (*ibuf[i])->audio->nb_samples - am->queue[i].pos);
+        /* Unroll the most common sample formats: speed +~350% for the loop,
+           +~13% overall (including two common decoders) */
+        switch (am->bps) {
+            case 1:
+                copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 1);
+                break;
+            case 2:
+                copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 2);
+                break;
+            case 4:
+                copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, 4);
+                break;
+            default:
+                copy_samples(am->nb_in_ch, am->route, ins, &outs, ns, am->bps);
+                break;
+        }
+
+        nb_samples -= ns;
+        for (i = 0; i < 2; i++) {
+            am->queue[i].nb_samples -= ns;
+            am->queue[i].pos += ns;
+            if (am->queue[i].pos == (*ibuf[i])->audio->nb_samples) {
+                am->queue[i].pos = 0;
+                avfilter_unref_buffer(*ibuf[i]);
+                *ibuf[i] = NULL;
+                ibuf[i]++;
+                ins[i] = *ibuf[i] ? (*ibuf[i])->data[0] : NULL;
+            }
+        }
+    }
+    for (i = 0; i < 2; i++) {
+        int nbufused = ibuf[i] - am->queue[i].buf;
+        if (nbufused) {
+            am->queue[i].nb_buf -= nbufused;
+            memmove(am->queue[i].buf, ibuf[i],
+                    am->queue[i].nb_buf * sizeof(**ibuf));
+        }
+    }
+    avfilter_filter_samples(ctx->outputs[0], outbuf);
+}
+
+AVFilter avfilter_af_amerge = {
+    .name          = "amerge",
+    .description   = NULL_IF_CONFIG_SMALL("Merge two audio streams into "
+                                          "a single multi-channel stream"),
+    .priv_size     = sizeof(struct amerge_context),
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+
+    .inputs    = (const AVFilterPad[]) {
+        { .name             = "in1",
+          .type             = AVMEDIA_TYPE_AUDIO,
+          .filter_samples   = filter_samples,
+          .min_perms        = AV_PERM_READ, },
+        { .name             = "in2",
+          .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_props,
+          .request_frame    = request_frame, },
+        { .name = NULL }
+    },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index bf17a7d..993e76d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -36,6 +36,7 @@ void avfilter_register_all(void)
 
     REGISTER_FILTER (ACONVERT,    aconvert,    af);
     REGISTER_FILTER (AFORMAT,     aformat,     af);
+    REGISTER_FILTER (AMERGE,      amerge,      af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);
-- 
1.7.2.5



More information about the ffmpeg-devel mailing list