237 lines
8.4 KiB
C
237 lines
8.4 KiB
C
/*
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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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* sample format and channel layout conversion audio filter
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*/
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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 "libavresample/avresample.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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typedef struct ResampleContext {
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AVAudioResampleContext *avr;
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int64_t next_pts;
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} ResampleContext;
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ResampleContext *s = ctx->priv;
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if (s->avr) {
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avresample_close(s->avr);
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avresample_free(&s->avr);
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}
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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AVFilterFormats *in_samplerates = ff_all_samplerates();
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AVFilterFormats *out_samplerates = ff_all_samplerates();
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AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
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AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
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ff_formats_ref(in_formats, &inlink->out_formats);
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ff_formats_ref(out_formats, &outlink->in_formats);
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ff_formats_ref(in_samplerates, &inlink->out_samplerates);
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ff_formats_ref(out_samplerates, &outlink->in_samplerates);
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ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
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ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
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return 0;
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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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AVFilterLink *inlink = ctx->inputs[0];
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ResampleContext *s = ctx->priv;
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char buf1[64], buf2[64];
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int ret;
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if (s->avr) {
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avresample_close(s->avr);
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avresample_free(&s->avr);
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}
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if (inlink->channel_layout == outlink->channel_layout &&
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inlink->sample_rate == outlink->sample_rate &&
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inlink->format == outlink->format)
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return 0;
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if (!(s->avr = avresample_alloc_context()))
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return AVERROR(ENOMEM);
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av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
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av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
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av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
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av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
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av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
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av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
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/* if both the input and output formats are s16 or u8, use s16 as
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the internal sample format */
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if (av_get_bytes_per_sample(inlink->format) <= 2 &&
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av_get_bytes_per_sample(outlink->format) <= 2)
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av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
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if ((ret = avresample_open(s->avr)) < 0)
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return ret;
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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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av_get_channel_layout_string(buf1, sizeof(buf1),
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-1, inlink ->channel_layout);
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av_get_channel_layout_string(buf2, sizeof(buf2),
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-1, outlink->channel_layout);
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av_log(ctx, AV_LOG_VERBOSE,
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"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
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av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
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av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
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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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ResampleContext *s = ctx->priv;
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int ret = ff_request_frame(ctx->inputs[0]);
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/* flush the lavr delay buffer */
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if (ret == AVERROR_EOF && s->avr) {
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AVFilterBufferRef *buf;
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int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
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outlink->sample_rate,
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ctx->inputs[0]->sample_rate,
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AV_ROUND_UP);
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if (!nb_samples)
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return ret;
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buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
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if (!buf)
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return AVERROR(ENOMEM);
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ret = avresample_convert(s->avr, (void**)buf->extended_data,
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buf->linesize[0], nb_samples,
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NULL, 0, 0);
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if (ret <= 0) {
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avfilter_unref_buffer(buf);
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return (ret == 0) ? AVERROR_EOF : ret;
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}
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buf->pts = s->next_pts;
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ff_filter_samples(outlink, buf);
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return 0;
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}
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return ret;
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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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ResampleContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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if (s->avr) {
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AVFilterBufferRef *buf_out;
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int delay, nb_samples, ret;
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/* maximum possible samples lavr can output */
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delay = avresample_get_delay(s->avr);
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nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
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outlink->sample_rate, inlink->sample_rate,
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AV_ROUND_UP);
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buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
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ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
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buf_out->linesize[0], nb_samples,
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(void**)buf->extended_data, buf->linesize[0],
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buf->audio->nb_samples);
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av_assert0(!avresample_available(s->avr));
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if (s->next_pts == AV_NOPTS_VALUE) {
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if (buf->pts == AV_NOPTS_VALUE) {
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av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
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"assuming 0.\n");
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s->next_pts = 0;
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} else
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s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
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outlink->time_base);
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}
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if (ret > 0) {
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buf_out->audio->nb_samples = ret;
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if (buf->pts != AV_NOPTS_VALUE) {
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buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
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outlink->time_base) -
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av_rescale(delay, outlink->sample_rate,
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inlink->sample_rate);
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} else
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buf_out->pts = s->next_pts;
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s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
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ff_filter_samples(outlink, buf_out);
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}
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avfilter_unref_buffer(buf);
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} else
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ff_filter_samples(outlink, buf);
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}
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AVFilter avfilter_af_resample = {
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.name = "resample",
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.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
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.priv_size = sizeof(ResampleContext),
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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.request_frame = request_frame },
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{ .name = NULL}},
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};
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