lavfi: add amerge audio filter.

Changelog

@@ -13,6 +13,7 @@ version next:
 - asplit audio filter
 - tinterlace video filter
 - astreamsync audio filter
+- amerge audio filter
 
 
 version 0.9:

doc/filters.texi

@@ -156,6 +156,39 @@ aformat=u8\\,s16:mono:packed
 aformat=s16:mono\\,stereo:all
 @end example
 
+@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.
+
+If inputs do not have the same duration, the output will stop with the
+shortest.
+
+Example: merge two mono files into a stereo stream:
+@example
+amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
+@end example
+
 @section anull
 
 Pass the audio source unchanged to the output.

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

libavfilter/af_amerge.c

+/*
+ * Copyright (c) 2011 Nicolas George <nicolas.george@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 "libswresample/swresample.h" // only for SWR_CH_MAX
+#include "avfilter.h"
+#include "internal.h"
+
+#define QUEUE_SIZE 16
+
+typedef struct {
+    int nb_in_ch[2];       /**< number of channels for each input */
+    int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */
+    int bps;
+    struct amerge_queue {
+        AVFilterBufferRef *buf[QUEUE_SIZE];
+        int nb_buf, nb_samples, pos;
+    } queue[2];
+} AMergeContext;
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AMergeContext *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)
+{
+    AMergeContext *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 (max %d)\n", SWR_CH_MAX);
+        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_ch_number = 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_ch_number++;
+    }
+    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_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AMergeContext *am = ctx->priv;
+    int64_t layout;
+    char name[3][256];
+    int i;
+
+    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);
+    outlink->sample_rate = ctx->inputs[0]->sample_rate;
+    outlink->time_base   = ctx->inputs[0]->time_base;
+    for (i = 0; i < 3; i++) {
+        layout = (i < 2 ? ctx->inputs[i] : ctx->outputs[0])->channel_layout;
+        av_get_channel_layout_string(name[i], sizeof(name[i]), -1, layout);
+    }
+    av_log(ctx, AV_LOG_INFO,
+           "in1:%s + in2:%s -> out:%s\n", name[0], name[1], name[2]);
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AMergeContext *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;
+}
+
+/**
+ * Copy samples from two input streams to one output stream.
+ * @param nb_in_ch  number of channels in each input stream
+ * @param route     routing values;
+ *                  input channel i goes to output channel route[i];
+ *                  i <  nb_in_ch[0] are the channels from the first output;
+ *                  i >= nb_in_ch[0] are the channels from the second output
+ * @param ins       pointer to the samples of each inputs, in packed format;
+ *                  will be left at the end of the copied samples
+ * @param outs      pointer to the samples of the output, in packet format;
+ *                  must point to a buffer big enough;
+ *                  will be left at the end of the copied samples
+ * @param ns        number of samples to copy
+ * @param bps       bytes per sample
+ */
+static inline void copy_samples(int nb_in_ch[2], int *route, uint8_t *ins[2],
+                                uint8_t **outs, int ns, int bps)
+{
+    int *route_cur;
+    int i, c;
+
+    while (ns--) {
+        route_cur = route;
+        for (i = 0; i < 2; i++) {
+            for (c = 0; c < nb_in_ch[i]; c++) {
+                memcpy((*outs) + bps * *(route_cur++), 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;
+    AMergeContext *am = ctx->priv;
+    int input_number = inlink == ctx->inputs[1];
+    struct amerge_queue *inq = &am->queue[input_number];
+    int nb_samples, ns, i;
+    AVFilterBufferRef *outbuf, **inbuf[2];
+    uint8_t *ins[2], *outs;
+
+    if (inq->nb_buf == QUEUE_SIZE) {
+        av_log(ctx, AV_LOG_ERROR, "Packet queue overflow; dropped\n");
+        avfilter_unref_buffer(insamples);
+        return;
+    }
+    inq->buf[inq->nb_buf++] = avfilter_ref_buffer(insamples, AV_PERM_READ |
+                                                             AV_PERM_PRESERVE);
+    inq->nb_samples += insamples->audio->nb_samples;
+    avfilter_unref_buffer(insamples);
+    if (!am->queue[!input_number].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++) {
+        inbuf[i] = am->queue[i].buf;
+        ins[i] = (*inbuf[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, (*inbuf[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 == (*inbuf[i])->audio->nb_samples) {
+                am->queue[i].pos = 0;
+                avfilter_unref_buffer(*inbuf[i]);
+                *inbuf[i] = NULL;
+                inbuf[i]++;
+                ins[i] = *inbuf[i] ? (*inbuf[i])->data[0] : NULL;
+            }
+        }
+    }
+    for (i = 0; i < 2; i++) {
+        int nbufused = inbuf[i] - am->queue[i].buf;
+        if (nbufused) {
+            am->queue[i].nb_buf -= nbufused;
+            memmove(am->queue[i].buf, inbuf[i],
+                    am->queue[i].nb_buf * sizeof(**inbuf));
+        }
+    }
+    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(AMergeContext),
+    .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_output,
+          .request_frame    = request_frame, },
+        { .name = NULL }
+    },
+};

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);

libavfilter/avfilter.h

@@ -30,7 +30,7 @@
 #include "libavcodec/avcodec.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 56
+#define LIBAVFILTER_VERSION_MINOR 57
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \