lavfi: add audio mix filter

Changelog

@@ -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:

doc/filters.texi

@@ -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.

libavfilter/Makefile

@@ -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

libavfilter/af_amix.c

+/*
+ * 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}},
+};

libavfilter/allfilters.c

@@ -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);

libavfilter/version.h

@@ -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, \