mirror of https://git.ffmpeg.org/ffmpeg.git
579 lines
18 KiB
C
579 lines
18 KiB
C
/*
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* Audio Mix Filter
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Audio Mix Filter
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*
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* Mixes audio from multiple sources into a single output. The channel layout,
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* sample rate, and sample format will be the same for all inputs and the
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* output.
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*/
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#include "libavutil/attributes.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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#define INPUT_ON 1 /**< input is active */
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#define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
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#define DURATION_LONGEST 0
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#define DURATION_SHORTEST 1
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#define DURATION_FIRST 2
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typedef struct FrameInfo {
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int nb_samples;
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int64_t pts;
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struct FrameInfo *next;
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} FrameInfo;
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/**
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* Linked list used to store timestamps and frame sizes of all frames in the
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* FIFO for the first input.
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*
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* This is needed to keep timestamps synchronized for the case where multiple
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* input frames are pushed to the filter for processing before a frame is
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* requested by the output link.
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*/
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typedef struct FrameList {
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int nb_frames;
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int nb_samples;
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FrameInfo *list;
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FrameInfo *end;
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} FrameList;
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static void frame_list_clear(FrameList *frame_list)
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{
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if (frame_list) {
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while (frame_list->list) {
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FrameInfo *info = frame_list->list;
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frame_list->list = info->next;
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av_free(info);
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}
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frame_list->nb_frames = 0;
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frame_list->nb_samples = 0;
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frame_list->end = NULL;
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}
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}
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static int frame_list_next_frame_size(FrameList *frame_list)
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{
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if (!frame_list->list)
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return 0;
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return frame_list->list->nb_samples;
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}
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static int64_t frame_list_next_pts(FrameList *frame_list)
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{
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if (!frame_list->list)
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return AV_NOPTS_VALUE;
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return frame_list->list->pts;
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}
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static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
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{
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if (nb_samples >= frame_list->nb_samples) {
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frame_list_clear(frame_list);
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} else {
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int samples = nb_samples;
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while (samples > 0) {
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FrameInfo *info = frame_list->list;
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av_assert0(info);
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if (info->nb_samples <= samples) {
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samples -= info->nb_samples;
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frame_list->list = info->next;
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if (!frame_list->list)
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frame_list->end = NULL;
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frame_list->nb_frames--;
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frame_list->nb_samples -= info->nb_samples;
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av_free(info);
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} else {
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info->nb_samples -= samples;
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info->pts += samples;
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frame_list->nb_samples -= samples;
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samples = 0;
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}
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}
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}
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}
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static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
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{
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FrameInfo *info = av_malloc(sizeof(*info));
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if (!info)
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return AVERROR(ENOMEM);
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info->nb_samples = nb_samples;
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info->pts = pts;
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info->next = NULL;
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if (!frame_list->list) {
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frame_list->list = info;
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frame_list->end = info;
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} else {
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av_assert0(frame_list->end);
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frame_list->end->next = info;
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frame_list->end = info;
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}
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frame_list->nb_frames++;
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frame_list->nb_samples += nb_samples;
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return 0;
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}
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typedef struct MixContext {
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const AVClass *class; /**< class for AVOptions */
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AVFloatDSPContext *fdsp;
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int nb_inputs; /**< number of inputs */
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int active_inputs; /**< number of input currently active */
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int duration_mode; /**< mode for determining duration */
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float dropout_transition; /**< transition time when an input drops out */
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int nb_channels; /**< number of channels */
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int sample_rate; /**< sample rate */
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int planar;
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AVAudioFifo **fifos; /**< audio fifo for each input */
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uint8_t *input_state; /**< current state of each input */
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float *input_scale; /**< mixing scale factor for each input */
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float scale_norm; /**< normalization factor for all inputs */
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int64_t next_pts; /**< calculated pts for next output frame */
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FrameList *frame_list; /**< list of frame info for the first input */
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} MixContext;
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#define OFFSET(x) offsetof(MixContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define F AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption amix_options[] = {
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{ "inputs", "Number of inputs.",
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OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
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{ "duration", "How to determine the end-of-stream.",
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OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
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{ "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A|F, "duration" },
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{ "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" },
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{ "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A|F, "duration" },
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{ "dropout_transition", "Transition time, in seconds, for volume "
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"renormalization when an input stream ends.",
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OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(amix);
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/**
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* Update the scaling factors to apply to each input during mixing.
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*
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* This balances the full volume range between active inputs and handles
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* volume transitions when EOF is encountered on an input but mixing continues
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* with the remaining inputs.
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*/
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static void calculate_scales(MixContext *s, int nb_samples)
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{
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int i;
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if (s->scale_norm > s->active_inputs) {
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s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
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s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);
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}
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for (i = 0; i < s->nb_inputs; i++) {
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if (s->input_state[i] & INPUT_ON)
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s->input_scale[i] = 1.0f / s->scale_norm;
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else
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s->input_scale[i] = 0.0f;
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}
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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MixContext *s = ctx->priv;
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int i;
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char buf[64];
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s->planar = av_sample_fmt_is_planar(outlink->format);
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s->sample_rate = outlink->sample_rate;
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outlink->time_base = (AVRational){ 1, outlink->sample_rate };
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s->next_pts = AV_NOPTS_VALUE;
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s->frame_list = av_mallocz(sizeof(*s->frame_list));
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if (!s->frame_list)
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return AVERROR(ENOMEM);
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s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
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if (!s->fifos)
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return AVERROR(ENOMEM);
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s->nb_channels = outlink->channels;
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for (i = 0; i < s->nb_inputs; i++) {
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s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
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if (!s->fifos[i])
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return AVERROR(ENOMEM);
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}
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s->input_state = av_malloc(s->nb_inputs);
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if (!s->input_state)
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return AVERROR(ENOMEM);
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memset(s->input_state, INPUT_ON, s->nb_inputs);
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s->active_inputs = s->nb_inputs;
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s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
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if (!s->input_scale)
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return AVERROR(ENOMEM);
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s->scale_norm = s->active_inputs;
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calculate_scales(s, 0);
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av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
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av_log(ctx, AV_LOG_VERBOSE,
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"inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
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av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
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return 0;
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}
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static int calc_active_inputs(MixContext *s);
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/**
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* Read samples from the input FIFOs, mix, and write to the output link.
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*/
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static int output_frame(AVFilterLink *outlink, int need_request)
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{
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AVFilterContext *ctx = outlink->src;
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MixContext *s = ctx->priv;
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AVFrame *out_buf, *in_buf;
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int nb_samples, ns, ret, i;
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ret = calc_active_inputs(s);
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if (ret < 0)
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return ret;
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if (s->input_state[0] & INPUT_ON) {
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/* first input live: use the corresponding frame size */
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nb_samples = frame_list_next_frame_size(s->frame_list);
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for (i = 1; i < s->nb_inputs; i++) {
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if (s->input_state[i] & INPUT_ON) {
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ns = av_audio_fifo_size(s->fifos[i]);
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if (ns < nb_samples) {
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if (!(s->input_state[i] & INPUT_EOF))
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/* unclosed input with not enough samples */
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return need_request ? ff_request_frame(ctx->inputs[i]) : 0;
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/* closed input to drain */
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nb_samples = ns;
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}
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}
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}
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} else {
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/* first input closed: use the available samples */
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nb_samples = INT_MAX;
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for (i = 1; i < s->nb_inputs; i++) {
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if (s->input_state[i] & INPUT_ON) {
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ns = av_audio_fifo_size(s->fifos[i]);
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nb_samples = FFMIN(nb_samples, ns);
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}
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}
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if (nb_samples == INT_MAX)
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return AVERROR_EOF;
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}
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s->next_pts = frame_list_next_pts(s->frame_list);
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frame_list_remove_samples(s->frame_list, nb_samples);
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calculate_scales(s, nb_samples);
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if (nb_samples == 0)
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return 0;
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out_buf = ff_get_audio_buffer(outlink, nb_samples);
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if (!out_buf)
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return AVERROR(ENOMEM);
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in_buf = ff_get_audio_buffer(outlink, nb_samples);
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if (!in_buf) {
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av_frame_free(&out_buf);
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return AVERROR(ENOMEM);
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}
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for (i = 0; i < s->nb_inputs; i++) {
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if (s->input_state[i] & INPUT_ON) {
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int planes, plane_size, p;
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av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
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nb_samples);
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planes = s->planar ? s->nb_channels : 1;
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plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
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plane_size = FFALIGN(plane_size, 16);
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if (out_buf->format == AV_SAMPLE_FMT_FLT ||
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out_buf->format == AV_SAMPLE_FMT_FLTP) {
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for (p = 0; p < planes; p++) {
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s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
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(float *) in_buf->extended_data[p],
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s->input_scale[i], plane_size);
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}
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} else {
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for (p = 0; p < planes; p++) {
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s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
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(double *) in_buf->extended_data[p],
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s->input_scale[i], plane_size);
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}
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}
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}
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}
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av_frame_free(&in_buf);
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out_buf->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += nb_samples;
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return ff_filter_frame(outlink, out_buf);
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}
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/**
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* Requests a frame, if needed, from each input link other than the first.
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*/
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static int request_samples(AVFilterContext *ctx, int min_samples)
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{
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MixContext *s = ctx->priv;
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int i, ret;
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av_assert0(s->nb_inputs > 1);
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for (i = 1; i < s->nb_inputs; i++) {
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ret = 0;
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if (!(s->input_state[i] & INPUT_ON))
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continue;
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if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
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continue;
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ret = ff_request_frame(ctx->inputs[i]);
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if (ret == AVERROR_EOF) {
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s->input_state[i] |= INPUT_EOF;
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if (av_audio_fifo_size(s->fifos[i]) == 0) {
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s->input_state[i] = 0;
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continue;
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}
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} else if (ret < 0)
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return ret;
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}
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return output_frame(ctx->outputs[0], 1);
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}
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|
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/**
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* Calculates the number of active inputs and determines EOF based on the
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* duration option.
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*
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* @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
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*/
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static int calc_active_inputs(MixContext *s)
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{
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int i;
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int active_inputs = 0;
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for (i = 0; i < s->nb_inputs; i++)
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active_inputs += !!(s->input_state[i] & INPUT_ON);
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s->active_inputs = active_inputs;
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|
|
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if (!active_inputs ||
|
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(s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
|
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(s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
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return AVERROR_EOF;
|
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return 0;
|
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}
|
|
|
|
static int request_frame(AVFilterLink *outlink)
|
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{
|
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AVFilterContext *ctx = outlink->src;
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MixContext *s = ctx->priv;
|
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int ret;
|
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int wanted_samples;
|
|
|
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ret = calc_active_inputs(s);
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if (ret < 0)
|
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return ret;
|
|
|
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if (!(s->input_state[0] & INPUT_ON))
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return request_samples(ctx, 1);
|
|
|
|
if (s->frame_list->nb_frames == 0) {
|
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ret = ff_request_frame(ctx->inputs[0]);
|
|
if (ret == AVERROR_EOF) {
|
|
s->input_state[0] = 0;
|
|
if (s->nb_inputs == 1)
|
|
return AVERROR_EOF;
|
|
return output_frame(ctx->outputs[0], 1);
|
|
}
|
|
return ret;
|
|
}
|
|
av_assert0(s->frame_list->nb_frames > 0);
|
|
|
|
wanted_samples = frame_list_next_frame_size(s->frame_list);
|
|
|
|
return request_samples(ctx, wanted_samples);
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
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MixContext *s = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
int i, ret = 0;
|
|
|
|
for (i = 0; i < ctx->nb_inputs; i++)
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if (ctx->inputs[i] == inlink)
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|
break;
|
|
if (i >= ctx->nb_inputs) {
|
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av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
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ret = AVERROR(EINVAL);
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goto fail;
|
|
}
|
|
|
|
if (i == 0) {
|
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int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
|
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outlink->time_base);
|
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ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
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if (ret < 0)
|
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goto fail;
|
|
}
|
|
|
|
ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
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buf->nb_samples);
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if (ret < 0)
|
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goto fail;
|
|
|
|
av_frame_free(&buf);
|
|
return output_frame(outlink, 0);
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|
|
|
fail:
|
|
av_frame_free(&buf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static av_cold int init(AVFilterContext *ctx)
|
|
{
|
|
MixContext *s = ctx->priv;
|
|
int i;
|
|
|
|
for (i = 0; i < s->nb_inputs; i++) {
|
|
char name[32];
|
|
AVFilterPad pad = { 0 };
|
|
|
|
snprintf(name, sizeof(name), "input%d", i);
|
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pad.type = AVMEDIA_TYPE_AUDIO;
|
|
pad.name = av_strdup(name);
|
|
if (!pad.name)
|
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return AVERROR(ENOMEM);
|
|
pad.filter_frame = filter_frame;
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|
|
|
ff_insert_inpad(ctx, i, &pad);
|
|
}
|
|
|
|
s->fdsp = avpriv_float_dsp_alloc(0);
|
|
if (!s->fdsp)
|
|
return AVERROR(ENOMEM);
|
|
|
|
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->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,
|
|
.request_frame = request_frame
|
|
},
|
|
{ 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,
|
|
.query_formats = query_formats,
|
|
.inputs = NULL,
|
|
.outputs = avfilter_af_amix_outputs,
|
|
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
|
|
};
|