/* * 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 * sample format and channel layout conversion audio filter */ #include "libavutil/avassert.h"#include "libavutil/avstring.h"#include "libavutil/common.h"#include "libavutil/dict.h"#include "libavutil/mathematics.h"#include "libavutil/opt.h" #include "libavresample/avresample.h" #include "audio.h"#include "avfilter.h"#include "formats.h"#include "internal.h" typedef struct ResampleContext {    const AVClass *class;    AVAudioResampleContext *avr;    AVDictionary *options;     int resampling;    int64_t next_pts;    int64_t next_in_pts;     /* set by filter_frame() to signal an output frame to request_frame() */    int got_output;} ResampleContext; static av_cold int init(AVFilterContext *ctx, AVDictionary **opts){    ResampleContext *s = ctx->priv;    const AVClass *avr_class = avresample_get_class();    AVDictionaryEntry *e = NULL;     while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {        if (av_opt_find(&avr_class, e->key, NULL, 0,                        AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))            av_dict_set(&s->options, e->key, e->value, 0);    }     e = NULL;    while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))        av_dict_set(opts, e->key, NULL, 0);     /* do not allow the user to override basic format options */    av_dict_set(&s->options,  "in_channel_layout", NULL, 0);    av_dict_set(&s->options, "out_channel_layout", NULL, 0);    av_dict_set(&s->options,  "in_sample_fmt",     NULL, 0);    av_dict_set(&s->options, "out_sample_fmt",     NULL, 0);    av_dict_set(&s->options,  "in_sample_rate",    NULL, 0);    av_dict_set(&s->options, "out_sample_rate",    NULL, 0);     return 0;} static av_cold void uninit(AVFilterContext *ctx){    ResampleContext *s = ctx->priv;     if (s->avr) {        avresample_close(s->avr);        avresample_free(&s->avr);    }    av_dict_free(&s->options);} static int query_formats(AVFilterContext *ctx){    AVFilterLink *inlink  = ctx->inputs[0];    AVFilterLink *outlink = ctx->outputs[0];    AVFilterFormats *in_formats, *out_formats, *in_samplerates, *out_samplerates;    AVFilterChannelLayouts *in_layouts, *out_layouts;    int ret;     if (!(in_formats      = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||        !(out_formats     = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||        !(in_samplerates  = ff_all_samplerates     (                  )) ||        !(out_samplerates = ff_all_samplerates     (                  )) ||        !(in_layouts      = ff_all_channel_layouts (                  )) ||        !(out_layouts     = ff_all_channel_layouts (                  )))        return AVERROR(ENOMEM);     if ((ret = ff_formats_ref         (in_formats,      &inlink->out_formats        )) < 0 ||        (ret = ff_formats_ref         (out_formats,     &outlink->in_formats        )) < 0 ||        (ret = ff_formats_ref         (in_samplerates,  &inlink->out_samplerates    )) < 0 ||        (ret = ff_formats_ref         (out_samplerates, &outlink->in_samplerates    )) < 0 ||        (ret = ff_channel_layouts_ref (in_layouts,      &inlink->out_channel_layouts)) < 0 ||        (ret = ff_channel_layouts_ref (out_layouts,     &outlink->in_channel_layouts)) < 0)        return ret;     return 0;} static int config_output(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    AVFilterLink *inlink = ctx->inputs[0];    ResampleContext   *s = ctx->priv;    char buf1[64], buf2[64];    int ret;     int64_t resampling_forced;     if (s->avr) {        avresample_close(s->avr);        avresample_free(&s->avr);    }     if (inlink->channel_layout == outlink->channel_layout &&        inlink->sample_rate    == outlink->sample_rate    &&        (inlink->format        == outlink->format ||        (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&         av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&         av_get_planar_sample_fmt(inlink->format) ==         av_get_planar_sample_fmt(outlink->format))))        return 0;     if (!(s->avr = avresample_alloc_context()))        return AVERROR(ENOMEM);     if (s->options) {        int ret;        AVDictionaryEntry *e = NULL;        while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))            av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);         ret = av_opt_set_dict(s->avr, &s->options);        if (ret < 0)            return ret;    }     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);    av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);    av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);     if ((ret = avresample_open(s->avr)) < 0)        return ret;     av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);    s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);     if (s->resampling) {        outlink->time_base = (AVRational){ 1, outlink->sample_rate };        s->next_pts        = AV_NOPTS_VALUE;        s->next_in_pts     = AV_NOPTS_VALUE;    } else        outlink->time_base = inlink->time_base;     av_get_channel_layout_string(buf1, sizeof(buf1),                                 -1, inlink ->channel_layout);    av_get_channel_layout_string(buf2, sizeof(buf2),                                 -1, outlink->channel_layout);    av_log(ctx, AV_LOG_VERBOSE,           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);     return 0;} static int request_frame(AVFilterLink *outlink){    AVFilterContext *ctx = outlink->src;    ResampleContext   *s = ctx->priv;    int ret = 0;     s->got_output = 0;    while (ret >= 0 && !s->got_output)        ret = ff_request_frame(ctx->inputs[0]);     /* flush the lavr delay buffer */    if (ret == AVERROR_EOF && s->avr) {        AVFrame *frame;        int nb_samples = avresample_get_out_samples(s->avr, 0);         if (!nb_samples)            return ret;         frame = ff_get_audio_buffer(outlink, nb_samples);        if (!frame)            return AVERROR(ENOMEM);         ret = avresample_convert(s->avr, frame->extended_data,                                 frame->linesize[0], nb_samples,                                 NULL, 0, 0);        if (ret <= 0) {            av_frame_free(&frame);            return (ret == 0) ? AVERROR_EOF : ret;        }         frame->nb_samples = ret;        frame->pts = s->next_pts;        return ff_filter_frame(outlink, frame);    }    return ret;} static int filter_frame(AVFilterLink *inlink, AVFrame *in){    AVFilterContext  *ctx = inlink->dst;    ResampleContext    *s = ctx->priv;    AVFilterLink *outlink = ctx->outputs[0];    int ret;     if (s->avr) {        AVFrame *out;        int delay, nb_samples;         /* maximum possible samples lavr can output */        delay      = avresample_get_delay(s->avr);        nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);         out = ff_get_audio_buffer(outlink, nb_samples);        if (!out) {            ret = AVERROR(ENOMEM);            goto fail;        }         ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],                                 nb_samples, in->extended_data, in->linesize[0],                                 in->nb_samples);        if (ret <= 0) {            av_frame_free(&out);            if (ret < 0)                goto fail;        }         av_assert0(!avresample_available(s->avr));         if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {            if (in->pts == AV_NOPTS_VALUE) {                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "                       "assuming 0.\n");                s->next_pts = 0;            } else                s->next_pts = av_rescale_q(in->pts, inlink->time_base,                                           outlink->time_base);        }         if (ret > 0) {            out->nb_samples = ret;             ret = av_frame_copy_props(out, in);            if (ret < 0) {                av_frame_free(&out);                goto fail;            }             if (s->resampling) {                out->sample_rate = outlink->sample_rate;                /* Only convert in->pts if there is a discontinuous jump.                   This ensures that out->pts tracks the number of samples actually                   output by the resampler in the absence of such a jump.                   Otherwise, the rounding in av_rescale_q() and av_rescale()                   causes off-by-1 errors. */                if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {                    out->pts = av_rescale_q(in->pts, inlink->time_base,                                                outlink->time_base) -                                   av_rescale(delay, outlink->sample_rate,                                              inlink->sample_rate);                } else                    out->pts = s->next_pts;                 s->next_pts = out->pts + out->nb_samples;                s->next_in_pts = in->pts + in->nb_samples;            } else                out->pts = in->pts;             ret = ff_filter_frame(outlink, out);            s->got_output = 1;        } fail:        av_frame_free(&in);    } else {        in->format = outlink->format;        ret = ff_filter_frame(outlink, in);        s->got_output = 1;    }     return ret;} static const AVClass *resample_child_class_next(const AVClass *prev){    return prev ? NULL : avresample_get_class();} static void *resample_child_next(void *obj, void *prev){    ResampleContext *s = obj;    return prev ? NULL : s->avr;} static const AVClass resample_class = {    .class_name       = "resample",    .item_name        = av_default_item_name,    .version          = LIBAVUTIL_VERSION_INT,    .child_class_next = resample_child_class_next,    .child_next       = resample_child_next,}; static const AVFilterPad avfilter_af_resample_inputs[] = {    {        .name          = "default",        .type          = AVMEDIA_TYPE_AUDIO,        .filter_frame  = filter_frame,    },    { NULL }}; static const AVFilterPad avfilter_af_resample_outputs[] = {    {        .name          = "default",        .type          = AVMEDIA_TYPE_AUDIO,        .config_props  = config_output,        .request_frame = request_frame    },    { NULL }}; AVFilter ff_af_resample = {    .name          = "resample",    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),    .priv_size     = sizeof(ResampleContext),    .priv_class    = &resample_class,    .init_dict     = init,    .uninit        = uninit,    .query_formats = query_formats,    .inputs        = avfilter_af_resample_inputs,    .outputs       = avfilter_af_resample_outputs,};