Signed-off-by: Paul B Mahol <one...@gmail.com> --- libavfilter/Makefile | 1 + libavfilter/af_aupsample.c | 159 +++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 3 files changed, 161 insertions(+) create mode 100644 libavfilter/af_aupsample.c
diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 682df45ef5..a38bc35231 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -86,6 +86,7 @@ OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o OBJS-$(CONFIG_ATRIM_FILTER) += trim.o +OBJS-$(CONFIG_AUPSAMPLE_FILTER) += af_aupsample.o OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o OBJS-$(CONFIG_BANDPASS_FILTER) += af_biquads.o OBJS-$(CONFIG_BANDREJECT_FILTER) += af_biquads.o diff --git a/libavfilter/af_aupsample.c b/libavfilter/af_aupsample.c new file mode 100644 index 0000000000..ee35b9c0c6 --- /dev/null +++ b/libavfilter/af_aupsample.c @@ -0,0 +1,159 @@ +/* + * Copyright (c) 2019 Paul B Mahol + * + * 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 + */ + +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" +#include "avfilter.h" +#include "audio.h" +#include "filters.h" +#include "internal.h" + +typedef struct AudioUpSampleContext { + const AVClass *class; + int factor; + + int64_t next_pts; +} AudioUpSampleContext; + +#define OFFSET(x) offsetof(AudioUpSampleContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption aupsample_options[] = { + { "factor", "set upsampling factor", OFFSET(factor), AV_OPT_TYPE_INT, {.i64=1}, 1, 64, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(aupsample); + +static int query_formats(AVFilterContext *ctx) +{ + AudioUpSampleContext *s = ctx->priv; + AVFilterChannelLayouts *layouts; + AVFilterFormats *formats; + int sample_rates[] = { 44100, -1 }; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + AVFilterFormats *avff; + int ret; + + if (!ctx->inputs[0]->in_samplerates || + !ctx->inputs[0]->in_samplerates->nb_formats) { + return AVERROR(EAGAIN); + } + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + avff = ctx->inputs[0]->in_samplerates; + sample_rates[0] = avff->formats[0]; + if (!ctx->inputs[0]->out_samplerates) + if ((ret = ff_formats_ref(ff_make_format_list(sample_rates), + &ctx->inputs[0]->out_samplerates)) < 0) + return ret; + + sample_rates[0] = avff->formats[0] * s->factor; + return ff_formats_ref(ff_make_format_list(sample_rates), + &ctx->outputs[0]->in_samplerates); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AudioUpSampleContext *s = ctx->priv; + + s->next_pts = AV_NOPTS_VALUE; + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + AudioUpSampleContext *s = ctx->priv; + const int factor = s->factor; + AVFrame *out; + + if (s->factor == 1) + return ff_filter_frame(outlink, in); + + out = ff_get_audio_buffer(outlink, in->nb_samples * s->factor); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + + if (s->next_pts == AV_NOPTS_VALUE) + s->next_pts = in->pts; + + for (int c = 0; c < in->channels; c++) { + const double *src = (const double *)in->extended_data[c]; + double *dst = (double *)out->extended_data[c]; + + for (int n = 0; n < in->nb_samples; n++) + dst[n*factor] = src[n]; + } + + out->pts = s->next_pts; + s->next_pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); + av_frame_free(&in); + return ff_filter_frame(ctx->outputs[0], out); +} + +static const AVFilterPad aupsample_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad aupsample_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_aupsample = { + .name = "aupsample", + .description = NULL_IF_CONFIG_SMALL("Upsample audio by integer factor."), + .query_formats = query_formats, + .priv_size = sizeof(AudioUpSampleContext), + .priv_class = &aupsample_class, + .inputs = aupsample_inputs, + .outputs = aupsample_outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 4d3039d6ba..29b372a1db 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -78,6 +78,7 @@ extern AVFilter ff_af_astats; extern AVFilter ff_af_astreamselect; extern AVFilter ff_af_atempo; extern AVFilter ff_af_atrim; +extern AVFilter ff_af_aupsample; extern AVFilter ff_af_azmq; extern AVFilter ff_af_bandpass; extern AVFilter ff_af_bandreject; -- 2.17.1 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".