/* * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * sample format and channel layout conversion audio filter */ #include "libavutil/avassert.h"#include "libavutil/avstring.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 {    AVAudioResampleContext *avr;     int64_t next_pts;} ResampleContext; static av_cold void uninit(AVFilterContext *ctx){    ResampleContext *s = ctx->priv;     if (s->avr) {        avresample_close(s->avr);        avresample_free(&s->avr);    }} static int query_formats(AVFilterContext *ctx){    AVFilterLink *inlink  = ctx->inputs[0];    AVFilterLink *outlink = ctx->outputs[0];     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);    AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);    AVFilterFormats        *in_samplerates  = ff_all_samplerates();    AVFilterFormats        *out_samplerates = ff_all_samplerates();    AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();    AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();     ff_formats_ref(in_formats,  &inlink->out_formats);    ff_formats_ref(out_formats, &outlink->in_formats);     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);    ff_formats_ref(out_samplerates, &outlink->in_samplerates);     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);     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;     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)        return 0;     if (!(s->avr = avresample_alloc_context()))        return AVERROR(ENOMEM);     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 both the input and output formats are s16 or u8, use s16 as       the internal sample format */    if (av_get_bytes_per_sample(inlink->format)  <= 2 &&        av_get_bytes_per_sample(outlink->format) <= 2)        av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);     if ((ret = avresample_open(s->avr)) < 0)        return ret;     outlink->time_base = (AVRational){ 1, outlink->sample_rate };    s->next_pts        = AV_NOPTS_VALUE;     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,#if FF_API_SAMPLERATE64           "fmt:%s srate:%"PRId64" cl:%s -> fmt:%s srate:%"PRId64" cl:%s\n",#else           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",#endif /* FF_API_SAMPLERATE64 */           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 = ff_request_frame(ctx->inputs[0]);     /* flush the lavr delay buffer */    if (ret == AVERROR_EOF && s->avr) {        AVFilterBufferRef *buf;        int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),                                        outlink->sample_rate,                                        ctx->inputs[0]->sample_rate,                                        AV_ROUND_UP);         if (!nb_samples)            return ret;         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);        if (!buf)            return AVERROR(ENOMEM);         ret = avresample_convert(s->avr, (void**)buf->extended_data,                                 buf->linesize[0], nb_samples,                                 NULL, 0, 0);        if (ret <= 0) {            avfilter_unref_buffer(buf);            return (ret == 0) ? AVERROR_EOF : ret;        }         buf->pts = s->next_pts;        ff_filter_samples(outlink, buf);        return 0;    }    return ret;} static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf){    AVFilterContext  *ctx = inlink->dst;    ResampleContext    *s = ctx->priv;    AVFilterLink *outlink = ctx->outputs[0];     if (s->avr) {        AVFilterBufferRef *buf_out;        int delay, nb_samples, ret;         /* maximum possible samples lavr can output */        delay      = avresample_get_delay(s->avr);        nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,                                    outlink->sample_rate, inlink->sample_rate,                                    AV_ROUND_UP);         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);        ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,                                     buf_out->linesize[0], nb_samples,                                     (void**)buf->extended_data, buf->linesize[0],                                     buf->audio->nb_samples);         av_assert0(!avresample_available(s->avr));         if (s->next_pts == AV_NOPTS_VALUE) {            if (buf->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(buf->pts, inlink->time_base,                                           outlink->time_base);        }         if (ret > 0) {            buf_out->audio->nb_samples = ret;            if (buf->pts != AV_NOPTS_VALUE) {                buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,                                            outlink->time_base) -                               av_rescale(delay, outlink->sample_rate,                                          inlink->sample_rate);            } else                buf_out->pts = s->next_pts;             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;             ff_filter_samples(outlink, buf_out);        }        avfilter_unref_buffer(buf);    } else        ff_filter_samples(outlink, buf);} AVFilter avfilter_af_resample = {    .name          = "resample",    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),    .priv_size     = sizeof(ResampleContext),     .uninit         = uninit,    .query_formats  = query_formats,     .inputs    = (const AVFilterPad[]) {{ .name            = "default",                                          .type            = AVMEDIA_TYPE_AUDIO,                                          .filter_samples  = filter_samples,                                          .min_perms       = AV_PERM_READ },                                        { .name = NULL}},    .outputs   = (const AVFilterPad[]) {{ .name          = "default",                                          .type          = AVMEDIA_TYPE_AUDIO,                                          .config_props  = config_output,                                          .request_frame = request_frame },                                        { .name = NULL}},};