| /* |
| * 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, |
| }; |