@@ -20,6 +20,7 @@ version <next>:

 - audio filters support in libavfilter and avconv

 - add fps filter

 - audio split filter

+- audio mix filter

 version 0.8:

@@ -133,6 +133,44 @@ For example to force the output to either unsigned 8-bit or signed 16-bit stereo

 aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo

 @end example

+@section amix

+

+Mixes multiple audio inputs into a single output.

+

+For example

+@example

+avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

+@end example

+will mix 3 input audio streams to a single output with the same duration as the

+first input and a dropout transition time of 3 seconds.

+

+The filter accepts the following named parameters:

+@table @option

+

+@item inputs

+Number of inputs. If unspecified, it defaults to 2.

+

+@item duration

+How to determine the end-of-stream.

+@table @option

+

+@item longest

+Duration of longest input. (default)

+

+@item shortest

+Duration of shortest input.

+

+@item first

+Duration of first input.

+

+@end table

+

+@item dropout_transition

+Transition time, in seconds, for volume renormalization when an input

+stream ends. The default value is 2 seconds.

+

+@end table

+

 @section anull

 Pass the audio source unchanged to the output.

@@ -25,6 +25,7 @@ OBJS = allfilters.o                                                     \

        video.o                                                          \

 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o

+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

new file mode 100644

@@ -0,0 +1,545 @@

+/*

+ * Audio Mix Filter

+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>

+ *

+ * 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

+ * Audio Mix Filter

+ *

+ * Mixes audio from multiple sources into a single output. The channel layout,

+ * sample rate, and sample format will be the same for all inputs and the

+ * output.

+ */

+

+#include "libavutil/audioconvert.h"

+#include "libavutil/audio_fifo.h"

+#include "libavutil/avassert.h"

+#include "libavutil/avstring.h"

+#include "libavutil/mathematics.h"

+#include "libavutil/opt.h"

+#include "libavutil/samplefmt.h"

+

+#include "audio.h"

+#include "avfilter.h"

+#include "formats.h"

+#include "internal.h"

+

+#define INPUT_OFF      0    /**< input has reached EOF */

+#define INPUT_ON       1    /**< input is active */

+#define INPUT_INACTIVE 2    /**< input is on, but is currently inactive */

+

+#define DURATION_LONGEST  0

+#define DURATION_SHORTEST 1

+#define DURATION_FIRST    2

+

+

+typedef struct FrameInfo {

+    int nb_samples;

+    int64_t pts;

+    struct FrameInfo *next;

+} FrameInfo;

+

+/**

+ * Linked list used to store timestamps and frame sizes of all frames in the

+ * FIFO for the first input.

+ *

+ * This is needed to keep timestamps synchronized for the case where multiple

+ * input frames are pushed to the filter for processing before a frame is

+ * requested by the output link.

+ */

+typedef struct FrameList {

+    int nb_frames;

+    int nb_samples;

+    FrameInfo *list;

+    FrameInfo *end;

+} FrameList;

+

+static void frame_list_clear(FrameList *frame_list)

+{

+    if (frame_list) {

+        while (frame_list->list) {

+            FrameInfo *info = frame_list->list;

+            frame_list->list = info->next;

+            av_free(info);

+        }

+        frame_list->nb_frames  = 0;

+        frame_list->nb_samples = 0;

+        frame_list->end        = NULL;

+    }

+}

+

+static int frame_list_next_frame_size(FrameList *frame_list)

+{

+    if (!frame_list->list)

+        return 0;

+    return frame_list->list->nb_samples;

+}

+

+static int64_t frame_list_next_pts(FrameList *frame_list)

+{

+    if (!frame_list->list)

+        return AV_NOPTS_VALUE;

+    return frame_list->list->pts;

+}

+

+static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)

+{

+    if (nb_samples >= frame_list->nb_samples) {

+        frame_list_clear(frame_list);

+    } else {

+        int samples = nb_samples;

+        while (samples > 0) {

+            FrameInfo *info = frame_list->list;

+            av_assert0(info != NULL);

+            if (info->nb_samples <= samples) {

+                samples -= info->nb_samples;

+                frame_list->list = info->next;

+                if (!frame_list->list)

+                    frame_list->end = NULL;

+                frame_list->nb_frames--;

+                frame_list->nb_samples -= info->nb_samples;

+                av_free(info);

+            } else {

+                info->nb_samples       -= samples;

+                info->pts              += samples;

+                frame_list->nb_samples -= samples;

+                samples = 0;

+            }

+        }

+    }

+}

+

+static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)

+{

+    FrameInfo *info = av_malloc(sizeof(*info));

+    if (!info)

+        return AVERROR(ENOMEM);

+    info->nb_samples = nb_samples;

+    info->pts        = pts;

+    info->next       = NULL;

+

+    if (!frame_list->list) {

+        frame_list->list = info;

+        frame_list->end  = info;

+    } else {

+        av_assert0(frame_list->end != NULL);

+        frame_list->end->next = info;

+        frame_list->end       = info;

+    }

+    frame_list->nb_frames++;

+    frame_list->nb_samples += nb_samples;

+

+    return 0;

+}

+

+

+typedef struct MixContext {

+    const AVClass *class;       /**< class for AVOptions */

+

+    int nb_inputs;              /**< number of inputs */

+    int active_inputs;          /**< number of input currently active */

+    int duration_mode;          /**< mode for determining duration */

+    float dropout_transition;   /**< transition time when an input drops out */

+

+    int nb_channels;            /**< number of channels */

+    int sample_rate;            /**< sample rate */

+    AVAudioFifo **fifos;        /**< audio fifo for each input */

+    uint8_t *input_state;       /**< current state of each input */

+    float *input_scale;         /**< mixing scale factor for each input */

+    float scale_norm;           /**< normalization factor for all inputs */

+    int64_t next_pts;           /**< calculated pts for next output frame */

+    FrameList *frame_list;      /**< list of frame info for the first input */

+} MixContext;

+

+#define OFFSET(x) offsetof(MixContext, x)

+#define A AV_OPT_FLAG_AUDIO_PARAM

+static const AVOption options[] = {

+    { "inputs", "Number of inputs.",

+            OFFSET(nb_inputs), AV_OPT_TYPE_INT, { 2 }, 1, 32, A },

+    { "duration", "How to determine the end-of-stream.",

+            OFFSET(duration_mode), AV_OPT_TYPE_INT, { DURATION_LONGEST }, 0,  2, A, "duration" },

+        { "longest",  "Duration of longest input.",  0, AV_OPT_TYPE_CONST, { DURATION_LONGEST  }, INT_MIN, INT_MAX, A, "duration" },

+        { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" },

+        { "first",    "Duration of first input.",    0, AV_OPT_TYPE_CONST, { DURATION_FIRST    }, INT_MIN, INT_MAX, A, "duration" },

+    { "dropout_transition", "Transition time, in seconds, for volume "

+                            "renormalization when an input stream ends.",

+            OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { 2.0 }, 0, INT_MAX, A },

+    { NULL },

+};

+

+static const AVClass amix_class = {

+    .class_name = "amix filter",

+    .item_name  = av_default_item_name,

+    .option     = options,

+    .version    = LIBAVUTIL_VERSION_INT,

+};

+

+

+/**

+ * Update the scaling factors to apply to each input during mixing.

+ *

+ * This balances the full volume range between active inputs and handles

+ * volume transitions when EOF is encountered on an input but mixing continues

+ * with the remaining inputs.

+ */

+static void calculate_scales(MixContext *s, int nb_samples)

+{

+    int i;

+

+    if (s->scale_norm > s->active_inputs) {

+        s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);

+        s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);

+    }

+

+    for (i = 0; i < s->nb_inputs; i++) {

+        if (s->input_state[i] == INPUT_ON)

+            s->input_scale[i] = 1.0f / s->scale_norm;

+        else

+            s->input_scale[i] = 0.0f;

+    }

+}

+

+static int config_output(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    MixContext *s      = ctx->priv;

+    int i;

+    char buf[64];

+

+    s->sample_rate     = outlink->sample_rate;

+    outlink->time_base = (AVRational){ 1, outlink->sample_rate };

+    s->next_pts        = AV_NOPTS_VALUE;

+

+    s->frame_list = av_mallocz(sizeof(*s->frame_list));

+    if (!s->frame_list)

+        return AVERROR(ENOMEM);

+

+    s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));

+    if (!s->fifos)

+        return AVERROR(ENOMEM);

+

+    s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);

+    for (i = 0; i < s->nb_inputs; i++) {

+        s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);

+        if (!s->fifos[i])

+            return AVERROR(ENOMEM);

+    }

+

+    s->input_state = av_malloc(s->nb_inputs);

+    if (!s->input_state)

+        return AVERROR(ENOMEM);

+    memset(s->input_state, INPUT_ON, s->nb_inputs);

+    s->active_inputs = s->nb_inputs;

+

+    s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));

+    if (!s->input_scale)

+        return AVERROR(ENOMEM);

+    s->scale_norm = s->active_inputs;

+    calculate_scales(s, 0);

+

+    av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);

+

+    av_log(ctx, AV_LOG_VERBOSE,

+           "inputs:%d fmt:%s srate:%"PRId64" cl:%s\n", s->nb_inputs,

+           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);

+

+    return 0;

+}

+

+/* TODO: move optimized version from DSPContext to libavutil */

+static void vector_fmac_scalar(float *dst, const float *src, float mul, int len)

+{

+    int i;

+    for (i = 0; i < len; i++)

+        dst[i] += src[i] * mul;

+}

+

+/**

+ * Read samples from the input FIFOs, mix, and write to the output link.

+ */

+static int output_frame(AVFilterLink *outlink, int nb_samples)

+{

+    AVFilterContext *ctx = outlink->src;

+    MixContext      *s = ctx->priv;

+    AVFilterBufferRef *out_buf, *in_buf;

+    int i;

+

+    calculate_scales(s, nb_samples);

+

+    out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

+    if (!out_buf)

+        return AVERROR(ENOMEM);

+

+    in_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

+    if (!in_buf)

+        return AVERROR(ENOMEM);

+

+    for (i = 0; i < s->nb_inputs; i++) {

+        if (s->input_state[i] == INPUT_ON) {

+            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,

+                               nb_samples);

+            vector_fmac_scalar((float *)out_buf->extended_data[0],

+                               (float *) in_buf->extended_data[0],

+                               s->input_scale[i], nb_samples * s->nb_channels);

+        }

+    }

+    avfilter_unref_buffer(in_buf);

+

+    out_buf->pts = s->next_pts;

+    if (s->next_pts != AV_NOPTS_VALUE)

+        s->next_pts += nb_samples;

+

+    ff_filter_samples(outlink, out_buf);

+

+    return 0;

+}

+

+/**

+ * Returns the smallest number of samples available in the input FIFOs other

+ * than that of the first input.

+ */

+static int get_available_samples(MixContext *s)

+{

+    int i;

+    int available_samples = INT_MAX;

+

+    av_assert0(s->nb_inputs > 1);

+

+    for (i = 1; i < s->nb_inputs; i++) {

+        int nb_samples;

+        if (s->input_state[i] == INPUT_OFF)

+            continue;

+        nb_samples = av_audio_fifo_size(s->fifos[i]);

+        available_samples = FFMIN(available_samples, nb_samples);

+    }

+    if (available_samples == INT_MAX)

+        return 0;

+    return available_samples;

+}

+

+/**

+ * Requests a frame, if needed, from each input link other than the first.

+ */

+static int request_samples(AVFilterContext *ctx, int min_samples)

+{

+    MixContext *s = ctx->priv;

+    int i, ret;

+

+    av_assert0(s->nb_inputs > 1);

+

+    for (i = 1; i < s->nb_inputs; i++) {

+        ret = 0;

+        if (s->input_state[i] == INPUT_OFF)

+            continue;

+        while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)

+            ret = avfilter_request_frame(ctx->inputs[i]);

+        if (ret == AVERROR_EOF) {

+            if (av_audio_fifo_size(s->fifos[i]) == 0) {

+                s->input_state[i] = INPUT_OFF;

+                continue;

+            }

+        } else if (ret)

+            return ret;

+    }

+    return 0;

+}

+

+/**

+ * Calculates the number of active inputs and determines EOF based on the

+ * duration option.

+ *

+ * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.

+ */

+static int calc_active_inputs(MixContext *s)

+{

+    int i;

+    int active_inputs = 0;

+    for (i = 0; i < s->nb_inputs; i++)

+        active_inputs += !!(s->input_state[i] != INPUT_OFF);

+    s->active_inputs = active_inputs;

+

+    if (!active_inputs ||

+        (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||

+        (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))

+        return AVERROR_EOF;

+    return 0;

+}

+

+static int request_frame(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    MixContext      *s = ctx->priv;

+    int ret;

+    int wanted_samples, available_samples;

+

+    if (s->input_state[0] == INPUT_OFF) {

+        ret = request_samples(ctx, 1);

+        if (ret < 0)

+            return ret;

+

+        ret = calc_active_inputs(s);

+        if (ret < 0)

+            return ret;

+

+        available_samples = get_available_samples(s);

+        if (!available_samples)

+            return 0;

+

+        return output_frame(outlink, available_samples);

+    }

+

+    if (s->frame_list->nb_frames == 0) {

+        ret = avfilter_request_frame(ctx->inputs[0]);

+        if (ret == AVERROR_EOF) {

+            s->input_state[0] = INPUT_OFF;

+            if (s->nb_inputs == 1)

+                return AVERROR_EOF;

+            else

+                return AVERROR(EAGAIN);

+        } else if (ret)

+            return ret;

+    }

+    av_assert0(s->frame_list->nb_frames > 0);

+

+    wanted_samples = frame_list_next_frame_size(s->frame_list);

+    ret = request_samples(ctx, wanted_samples);

+    if (ret < 0)

+        return ret;

+

+    ret = calc_active_inputs(s);

+    if (ret < 0)

+        return ret;

+

+    if (s->active_inputs > 1) {

+        available_samples = get_available_samples(s);

+        if (!available_samples)

+            return 0;

+        available_samples = FFMIN(available_samples, wanted_samples);

+    } else {

+        available_samples = wanted_samples;

+    }

+

+    s->next_pts = frame_list_next_pts(s->frame_list);

+    frame_list_remove_samples(s->frame_list, available_samples);

+

+    return output_frame(outlink, available_samples);

+}

+

+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)

+{

+    AVFilterContext  *ctx = inlink->dst;

+    MixContext       *s = ctx->priv;

+    AVFilterLink *outlink = ctx->outputs[0];

+    int i;

+

+    for (i = 0; i < ctx->input_count; i++)

+        if (ctx->inputs[i] == inlink)

+            break;

+    if (i >= ctx->input_count) {

+        av_log(ctx, AV_LOG_ERROR, "unknown input link\n");

+        return;

+    }

+

+    if (i == 0) {

+        int64_t pts = av_rescale_q(buf->pts, inlink->time_base,

+                                   outlink->time_base);

+        frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);

+    }

+

+    av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,

+                        buf->audio->nb_samples);

+

+    avfilter_unref_buffer(buf);

+}

+

+static int init(AVFilterContext *ctx, const char *args, void *opaque)

+{

+    MixContext *s = ctx->priv;

+    int i, ret;

+

+    s->class = &amix_class;

+    av_opt_set_defaults(s);

+

+    if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {

+        av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);

+        return ret;

+    }

+    av_opt_free(s);

+

+    for (i = 0; i < s->nb_inputs; i++) {

+        char name[32];

+        AVFilterPad pad = { 0 };

+

+        snprintf(name, sizeof(name), "input%d", i);

+        pad.type           = AVMEDIA_TYPE_AUDIO;

+        pad.name           = av_strdup(name);

+        pad.filter_samples = filter_samples;

+

+        avfilter_insert_inpad(ctx, i, &pad);

+    }

+

+    return 0;

+}

+

+static void uninit(AVFilterContext *ctx)

+{

+    int i;

+    MixContext *s = ctx->priv;

+

+    if (s->fifos) {

+        for (i = 0; i < s->nb_inputs; i++)

+            av_audio_fifo_free(s->fifos[i]);

+        av_freep(&s->fifos);

+    }

+    frame_list_clear(s->frame_list);

+    av_freep(&s->frame_list);

+    av_freep(&s->input_state);

+    av_freep(&s->input_scale);

+

+    for (i = 0; i < ctx->input_count; i++)

+        av_freep(&ctx->input_pads[i].name);

+}

+

+static int query_formats(AVFilterContext *ctx)

+{

+    AVFilterFormats *formats = NULL;

+    avfilter_add_format(&formats, AV_SAMPLE_FMT_FLT);

+    avfilter_set_common_formats(ctx, formats);

+    ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());

+    ff_set_common_samplerates(ctx, ff_all_samplerates());

+    return 0;

+}

+

+AVFilter avfilter_af_amix = {

+    .name          = "amix",

+    .description   = NULL_IF_CONFIG_SMALL("Audio mixing."),

+    .priv_size     = sizeof(MixContext),

+

+    .init           = init,

+    .uninit         = uninit,

+    .query_formats  = query_formats,

+

+    .inputs    = (const AVFilterPad[]) {{ .name = NULL}},

+    .outputs   = (const AVFilterPad[]) {{ .name          = "default",

+                                          .type          = AVMEDIA_TYPE_AUDIO,

+                                          .config_props  = config_output,

+                                          .request_frame = request_frame },

+                                        { .name = NULL}},

+};

@@ -35,6 +35,7 @@ void avfilter_register_all(void)

     initialized = 1;

     REGISTER_FILTER (AFORMAT,     aformat,     af);

+    REGISTER_FILTER (AMIX,        amix,        af);

     REGISTER_FILTER (ANULL,       anull,       af);

     REGISTER_FILTER (ASPLIT,      asplit,      af);

     REGISTER_FILTER (ASYNCTS,     asyncts,     af);

@@ -29,7 +29,7 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  2

-#define LIBAVFILTER_VERSION_MINOR  19

+#define LIBAVFILTER_VERSION_MINOR  20

 #define LIBAVFILTER_VERSION_MICRO  0

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \