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FFmpeg/libavfilter/af_amix.c

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/*
* Audio Mix Filter
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* 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
* Lesser 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 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/attributes.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "internal.h"
#define INPUT_ON 1 /**< input is active */
#define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
#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);
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);
frame_list->end->next = info;
frame_list->end = info;
}
frame_list->nb_frames++;
frame_list->nb_samples += nb_samples;
return 0;
}
/* FIXME: use directly links fifo */
typedef struct MixContext {
const AVClass *class; /**< class for AVOptions */
AVFloatDSPContext *fdsp;
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 */
char *weights_str; /**< string for custom weights for every input */
int nb_channels; /**< number of channels */
int sample_rate; /**< sample rate */
int planar;
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 *weights; /**< custom weights for every input */
float weight_sum; /**< sum of custom weights for every input */
float *scale_norm; /**< normalization factor for every input */
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
#define F AV_OPT_FLAG_FILTERING_PARAM
static const AVOption amix_options[] = {
{ "inputs", "Number of inputs.",
OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
{ "duration", "How to determine the end-of-stream.",
OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
{ "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
{ "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
{ "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
{ "dropout_transition", "Transition time, in seconds, for volume "
"renormalization when an input stream ends.",
OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
{ "weights", "Set weight for each input.",
OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F },
{ NULL }
};
AVFILTER_DEFINE_CLASS(amix);
/**
* 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)
{
float weight_sum = 0.f;
int i;
for (i = 0; i < s->nb_inputs; i++)
if (s->input_state[i] & INPUT_ON)
weight_sum += FFABS(s->weights[i]);
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
if (s->scale_norm[i] > weight_sum / FFABS(s->weights[i])) {
s->scale_norm[i] -= ((s->weight_sum / FFABS(s->weights[i])) / s->nb_inputs) *
nb_samples / (s->dropout_transition * s->sample_rate);
s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / FFABS(s->weights[i]));
}
}
}
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON)
s->input_scale[i] = 1.0f / s->scale_norm[i] * FFSIGN(s->weights[i]);
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->planar = av_sample_fmt_is_planar(outlink->format);
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_array(s->nb_inputs, sizeof(*s->fifos));
if (!s->fifos)
return AVERROR(ENOMEM);
s->nb_channels = outlink->channels;
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_array(s->nb_inputs, sizeof(*s->input_scale));
s->scale_norm = av_mallocz_array(s->nb_inputs, sizeof(*s->scale_norm));
if (!s->input_scale || !s->scale_norm)
return AVERROR(ENOMEM);
for (i = 0; i < s->nb_inputs; i++)
s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
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:%d cl:%s\n", s->nb_inputs,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
return 0;
}
/**
* Read samples from the input FIFOs, mix, and write to the output link.
*/
static int output_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
AVFrame *out_buf, *in_buf;
int nb_samples, ns, i;
if (s->input_state[0] & INPUT_ON) {
/* first input live: use the corresponding frame size */
nb_samples = frame_list_next_frame_size(s->frame_list);
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
if (ns < nb_samples) {
if (!(s->input_state[i] & INPUT_EOF))
/* unclosed input with not enough samples */
return 0;
/* closed input to drain */
nb_samples = ns;
}
}
}
} else {
/* first input closed: use the available samples */
nb_samples = INT_MAX;
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
nb_samples = FFMIN(nb_samples, ns);
}
}
if (nb_samples == INT_MAX) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, nb_samples);
calculate_scales(s, nb_samples);
if (nb_samples == 0)
return 0;
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
in_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!in_buf) {
av_frame_free(&out_buf);
return AVERROR(ENOMEM);
}
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
int planes, plane_size, p;
av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
nb_samples);
planes = s->planar ? s->nb_channels : 1;
plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
plane_size = FFALIGN(plane_size, 16);
if (out_buf->format == AV_SAMPLE_FMT_FLT ||
out_buf->format == AV_SAMPLE_FMT_FLTP) {
for (p = 0; p < planes; p++) {
s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
(float *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
} else {
for (p = 0; p < planes; p++) {
s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
(double *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
}
}
}
av_frame_free(&in_buf);
out_buf->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
return ff_filter_frame(outlink, out_buf);
}
/**
* 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;
av_assert0(s->nb_inputs > 1);
for (i = 1; i < s->nb_inputs; i++) {
if (!(s->input_state[i] & INPUT_ON) ||
(s->input_state[i] & INPUT_EOF))
continue;
if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
continue;
ff_inlink_request_frame(ctx->inputs[i]);
}
return output_frame(ctx->outputs[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_ON);
s->active_inputs = active_inputs;
if (!active_inputs ||
(s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
(s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
return AVERROR_EOF;
return 0;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *outlink = ctx->outputs[0];
MixContext *s = ctx->priv;
AVFrame *buf = NULL;
int i, ret;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
for (i = 0; i < s->nb_inputs; i++) {
AVFilterLink *inlink = ctx->inputs[i];
if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
if (i == 0) {
int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
if (ret < 0) {
av_frame_free(&buf);
return ret;
}
}
ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
buf->nb_samples);
if (ret < 0) {
av_frame_free(&buf);
return ret;
}
av_frame_free(&buf);
ret = output_frame(outlink);
if (ret < 0)
return ret;
}
}
for (i = 0; i < s->nb_inputs; i++) {
int64_t pts;
int status;
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
if (status == AVERROR_EOF) {
if (i == 0) {
s->input_state[i] = 0;
if (s->nb_inputs == 1) {
ff_outlink_set_status(outlink, status, pts);
return 0;
}
} else {
s->input_state[i] |= INPUT_EOF;
if (av_audio_fifo_size(s->fifos[i]) == 0) {
s->input_state[i] = 0;
}
}
}
}
}
if (calc_active_inputs(s)) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
if (ff_outlink_frame_wanted(outlink)) {
int wanted_samples;
if (!(s->input_state[0] & INPUT_ON))
return request_samples(ctx, 1);
if (s->frame_list->nb_frames == 0) {
ff_inlink_request_frame(ctx->inputs[0]);
return 0;
}
av_assert0(s->frame_list->nb_frames > 0);
wanted_samples = frame_list_next_frame_size(s->frame_list);
return request_samples(ctx, wanted_samples);
}
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
MixContext *s = ctx->priv;
char *p, *arg, *saveptr = NULL;
float last_weight = 1.f;
int i, ret;
for (i = 0; i < s->nb_inputs; i++) {
AVFilterPad pad = { 0 };
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_asprintf("input%d", i);
if (!pad.name)
return AVERROR(ENOMEM);
if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
av_freep(&pad.name);
return ret;
}
}
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
if (!s->weights)
return AVERROR(ENOMEM);
p = s->weights_str;
for (i = 0; i < s->nb_inputs; i++) {
if (!(arg = av_strtok(p, " ", &saveptr)))
break;
p = NULL;
sscanf(arg, "%f", &last_weight);
s->weights[i] = last_weight;
s->weight_sum += FFABS(last_weight);
}
for (; i < s->nb_inputs; i++) {
s->weights[i] = last_weight;
s->weight_sum += FFABS(last_weight);
}
return 0;
}
static av_cold 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);
av_freep(&s->scale_norm);
av_freep(&s->weights);
av_freep(&s->fdsp);
for (i = 0; i < ctx->nb_inputs; i++)
av_freep(&ctx->input_pads[i].name);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts;
int ret;
layouts = ff_all_channel_counts();
if (!layouts) {
ret = AVERROR(ENOMEM);
goto fail;
}
if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
(ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
(ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
(ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBLP)) < 0 ||
(ret = ff_set_common_formats (ctx, formats)) < 0 ||
(ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
(ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
goto fail;
return 0;
fail:
if (layouts)
av_freep(&layouts->channel_layouts);
av_freep(&layouts);
return ret;
}
static const AVFilterPad avfilter_af_amix_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};
AVFilter ff_af_amix = {
.name = "amix",
.description = NULL_IF_CONFIG_SMALL("Audio mixing."),
.priv_size = sizeof(MixContext),
.priv_class = &amix_class,
.init = init,
.uninit = uninit,
.activate = activate,
.query_formats = query_formats,
.inputs = NULL,
.outputs = avfilter_af_amix_outputs,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
};
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