@@ -13,6 +13,7 @@ version next:

 - asplit audio filter

 - tinterlace video filter

 - astreamsync audio filter

+- amerge audio filter

 version 0.9:

@@ -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.

@@ -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

new file mode 100644

@@ -0,0 +1,288 @@

+/*

+ * 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 }

+    },

+};

@@ -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);

@@ -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, \