/* * 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 "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; } 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 */ 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 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 #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, 32, 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 }, INT_MIN, INT_MAX, A|F, "duration" }, { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" }, { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, 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 }, { 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) { 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->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 = 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_array(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:%d cl:%s\n", s->nb_inputs, av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf); return 0; } static int calc_active_inputs(MixContext *s); /** * 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, ret, i; ret = calc_active_inputs(s); if (ret < 0) return ret; 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) return AVERROR_EOF; } s->next_pts = frame_list_next_pts(s->frame_list); frame_list_remove_samples(s->frame_list, nb_samples); calculate_scales(s, nb_samples); 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); 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); } } } 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, ret; av_assert0(s->nb_inputs > 1); for (i = 1; i < s->nb_inputs; i++) { ret = 0; if (!(s->input_state[i] & INPUT_ON)) continue; if (av_audio_fifo_size(s->fifos[i]) >= min_samples) continue; ret = ff_request_frame(ctx->inputs[i]); if (ret == AVERROR_EOF) { s->input_state[i] |= INPUT_EOF; if (av_audio_fifo_size(s->fifos[i]) == 0) { s->input_state[i] = 0; continue; } } else if (ret < 0) return ret; } 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 request_frame(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; MixContext *s = ctx->priv; int ret; int wanted_samples; ret = calc_active_inputs(s); if (ret < 0) return ret; if (!(s->input_state[0] & INPUT_ON)) return request_samples(ctx, 1); if (s->frame_list->nb_frames == 0) { 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]); } 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; MixContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; int i, ret = 0; for (i = 0; i < ctx->nb_inputs; i++) if (ctx->inputs[i] == inlink) break; if (i >= ctx->nb_inputs) { av_log(ctx, AV_LOG_ERROR, "unknown input link\n"); ret = AVERROR(EINVAL); goto fail; } 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) goto fail; } ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, buf->nb_samples); av_frame_free(&buf); return output_frame(outlink); 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); pad.type = AVMEDIA_TYPE_AUDIO; pad.name = av_strdup(name); if (!pad.name) return AVERROR(ENOMEM); pad.filter_frame = filter_frame; 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_layouts(); if (!layouts) return AVERROR(ENOMEM); if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT)) < 0) return ret; if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0) return ret; ret = ff_set_common_formats(ctx, formats); if (ret < 0) return ret; ret = ff_set_common_channel_layouts(ctx, layouts); if (ret < 0) return ret; return ff_set_common_samplerates(ctx, ff_all_samplerates()); } 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, };