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lavfi: add audio mix filter

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This commit is contained in:
Justin Ruggles 2012-05-21 21:27:59 -04:00
parent 1e8561e369
commit c7448c182a
6 changed files with 587 additions and 1 deletions
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1
Changelog
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@ -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:

38
doc/filters.texi
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@ -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.

1
libavfilter/Makefile
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@ -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

545
libavfilter/af_amix.c Normal file
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@ -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}},
};

1
libavfilter/allfilters.c
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@ -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);

2
libavfilter/version.h
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@ -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, \