mirror of https://github.com/FFmpeg/FFmpeg.git
Justin Ruggles
13 years ago
parent
1e8561e369
commit
c7448c182a
6 changed files with 587 additions and 1 deletions
@ -0,0 +1,545 @@ |
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/*
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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 Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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/audioconvert.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/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_OFF 0 /**< input has reached EOF */ |
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#define INPUT_ON 1 /**< input is active */ |
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#define INPUT_INACTIVE 2 /**< input is on, but is currently inactive */ |
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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 != NULL); |
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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 != NULL); |
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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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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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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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static const AVOption options[] = { |
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{ "inputs", "Number of inputs.", |
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OFFSET(nb_inputs), AV_OPT_TYPE_INT, { 2 }, 1, 32, A }, |
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{ "duration", "How to determine the end-of-stream.", |
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OFFSET(duration_mode), AV_OPT_TYPE_INT, { DURATION_LONGEST }, 0, 2, A, "duration" }, |
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{ "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { DURATION_LONGEST }, INT_MIN, INT_MAX, A, "duration" }, |
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{ "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" }, |
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{ "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { DURATION_FIRST }, INT_MIN, INT_MAX, A, "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, { 2.0 }, 0, INT_MAX, A }, |
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{ NULL }, |
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}; |
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static const AVClass amix_class = { |
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.class_name = "amix filter", |
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.item_name = av_default_item_name, |
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.option = options, |
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.version = LIBAVUTIL_VERSION_INT, |
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}; |
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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->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(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 = av_get_channel_layout_nb_channels(outlink->channel_layout); |
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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(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:%"PRId64" 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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/* TODO: move optimized version from DSPContext to libavutil */ |
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static void vector_fmac_scalar(float *dst, const float *src, float mul, int len) |
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{ |
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int i; |
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for (i = 0; i < len; i++) |
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dst[i] += src[i] * mul; |
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} |
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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 nb_samples) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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MixContext *s = ctx->priv; |
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AVFilterBufferRef *out_buf, *in_buf; |
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int i; |
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calculate_scales(s, nb_samples); |
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out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, 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, AV_PERM_WRITE, nb_samples); |
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if (!in_buf) |
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return AVERROR(ENOMEM); |
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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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av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data, |
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nb_samples); |
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vector_fmac_scalar((float *)out_buf->extended_data[0], |
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(float *) in_buf->extended_data[0], |
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s->input_scale[i], nb_samples * s->nb_channels); |
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} |
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} |
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avfilter_unref_buffer(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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ff_filter_samples(outlink, out_buf); |
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return 0; |
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} |
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/**
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* Returns the smallest number of samples available in the input FIFOs other |
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* than that of the first input. |
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*/ |
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static int get_available_samples(MixContext *s) |
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{ |
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int i; |
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int available_samples = INT_MAX; |
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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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int nb_samples; |
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if (s->input_state[i] == INPUT_OFF) |
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continue; |
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nb_samples = av_audio_fifo_size(s->fifos[i]); |
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available_samples = FFMIN(available_samples, nb_samples); |
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} |
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if (available_samples == INT_MAX) |
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return 0; |
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return available_samples; |
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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_OFF) |
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continue; |
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while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples) |
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ret = avfilter_request_frame(ctx->inputs[i]); |
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if (ret == AVERROR_EOF) { |
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if (av_audio_fifo_size(s->fifos[i]) == 0) { |
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s->input_state[i] = INPUT_OFF; |
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continue; |
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} |
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} else if (ret) |
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return ret; |
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} |
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return 0; |
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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_OFF); |
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s->active_inputs = active_inputs; |
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if (!active_inputs || |
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(s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) || |
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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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} |
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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, available_samples; |
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if (s->input_state[0] == INPUT_OFF) { |
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ret = request_samples(ctx, 1); |
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if (ret < 0) |
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return ret; |
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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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available_samples = get_available_samples(s); |
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if (!available_samples) |
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return 0; |
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return output_frame(outlink, available_samples); |
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} |
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if (s->frame_list->nb_frames == 0) { |
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ret = avfilter_request_frame(ctx->inputs[0]); |
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if (ret == AVERROR_EOF) { |
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s->input_state[0] = INPUT_OFF; |
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if (s->nb_inputs == 1) |
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return AVERROR_EOF; |
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else |
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return AVERROR(EAGAIN); |
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} else if (ret) |
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return ret; |
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} |
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av_assert0(s->frame_list->nb_frames > 0); |
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wanted_samples = frame_list_next_frame_size(s->frame_list); |
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ret = request_samples(ctx, wanted_samples); |
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if (ret < 0) |
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return ret; |
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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->active_inputs > 1) { |
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available_samples = get_available_samples(s); |
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if (!available_samples) |
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return 0; |
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available_samples = FFMIN(available_samples, wanted_samples); |
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} else { |
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available_samples = wanted_samples; |
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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, available_samples); |
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return output_frame(outlink, available_samples); |
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} |
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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MixContext *s = ctx->priv; |
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AVFilterLink *outlink = ctx->outputs[0]; |
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int i; |
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for (i = 0; i < ctx->input_count; i++) |
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if (ctx->inputs[i] == inlink) |
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break; |
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if (i >= ctx->input_count) { |
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av_log(ctx, AV_LOG_ERROR, "unknown input link\n"); |
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return; |
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} |
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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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frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts); |
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} |
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av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, |
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buf->audio->nb_samples); |
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avfilter_unref_buffer(buf); |
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} |
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static int init(AVFilterContext *ctx, const char *args, void *opaque) |
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{ |
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MixContext *s = ctx->priv; |
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int i, ret; |
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s->class = &amix_class; |
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av_opt_set_defaults(s); |
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if ((ret = av_set_options_string(s, args, "=", ":")) < 0) { |
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av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args); |
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return ret; |
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} |
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av_opt_free(s); |
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for (i = 0; i < s->nb_inputs; i++) { |
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char name[32]; |
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AVFilterPad pad = { 0 }; |
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snprintf(name, sizeof(name), "input%d", i); |
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pad.type = AVMEDIA_TYPE_AUDIO; |
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pad.name = av_strdup(name); |
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pad.filter_samples = filter_samples; |
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avfilter_insert_inpad(ctx, i, &pad); |
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} |
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return 0; |
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} |
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static void uninit(AVFilterContext *ctx) |
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{ |
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int i; |
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MixContext *s = ctx->priv; |
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if (s->fifos) { |
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for (i = 0; i < s->nb_inputs; i++) |
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av_audio_fifo_free(s->fifos[i]); |
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av_freep(&s->fifos); |
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} |
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frame_list_clear(s->frame_list); |
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av_freep(&s->frame_list); |
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av_freep(&s->input_state); |
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av_freep(&s->input_scale); |
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for (i = 0; i < ctx->input_count; i++) |
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av_freep(&ctx->input_pads[i].name); |
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} |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterFormats *formats = NULL; |
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avfilter_add_format(&formats, AV_SAMPLE_FMT_FLT); |
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avfilter_set_common_formats(ctx, formats); |
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ff_set_common_channel_layouts(ctx, ff_all_channel_layouts()); |
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ff_set_common_samplerates(ctx, ff_all_samplerates()); |
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return 0; |
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} |
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|
||||
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}}, |
||||
}; |
||||
Loading…
Reference in new issue