diff options
Diffstat (limited to 'libavfilter/af_aresample.c')
| -rw-r--r-- | libavfilter/af_aresample.c | 351 |
1 files changed, 351 insertions, 0 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c new file mode 100644 index 0000000000..028e105318 --- /dev/null +++ b/libavfilter/af_aresample.c @@ -0,0 +1,351 @@ +/* + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2011 Mina Nagy Zaki + * + * 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 + * resampling audio filter + */ + +#include "libavutil/avstring.h" +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" +#include "libavutil/avassert.h" +#include "libswresample/swresample.h" +#include "avfilter.h" +#include "audio.h" +#include "internal.h" + +typedef struct { + const AVClass *class; + int sample_rate_arg; + double ratio; + struct SwrContext *swr; + int64_t next_pts; + int more_data; +} AResampleContext; + +static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts) +{ + AResampleContext *aresample = ctx->priv; + int ret = 0; + + aresample->next_pts = AV_NOPTS_VALUE; + aresample->swr = swr_alloc(); + if (!aresample->swr) { + ret = AVERROR(ENOMEM); + goto end; + } + + if (opts) { + AVDictionaryEntry *e = NULL; + + while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) { + if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0) + goto end; + } + av_dict_free(opts); + } + if (aresample->sample_rate_arg > 0) + av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0); +end: + return ret; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AResampleContext *aresample = ctx->priv; + swr_free(&aresample->swr); +} + +static int query_formats(AVFilterContext *ctx) +{ + AResampleContext *aresample = ctx->priv; + enum AVSampleFormat out_format; + int64_t out_rate, out_layout; + + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + + AVFilterFormats *in_formats, *out_formats; + AVFilterFormats *in_samplerates, *out_samplerates; + AVFilterChannelLayouts *in_layouts, *out_layouts; + int ret; + + av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format); + av_opt_get_int(aresample->swr, "osr", 0, &out_rate); + av_opt_get_int(aresample->swr, "ocl", 0, &out_layout); + + in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); + if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0) + return ret; + + in_samplerates = ff_all_samplerates(); + if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0) + return ret; + + in_layouts = ff_all_channel_counts(); + if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0) + return ret; + + if(out_rate > 0) { + int ratelist[] = { out_rate, -1 }; + out_samplerates = ff_make_format_list(ratelist); + } else { + out_samplerates = ff_all_samplerates(); + } + + if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0) + return ret; + + if(out_format != AV_SAMPLE_FMT_NONE) { + int formatlist[] = { out_format, -1 }; + out_formats = ff_make_format_list(formatlist); + } else + out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); + if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0) + return ret; + + if(out_layout) { + int64_t layout_list[] = { out_layout, -1 }; + out_layouts = avfilter_make_format64_list(layout_list); + } else + out_layouts = ff_all_channel_counts(); + + return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts); +} + + +static int config_output(AVFilterLink *outlink) +{ + int ret; + AVFilterContext *ctx = outlink->src; + AVFilterLink *inlink = ctx->inputs[0]; + AResampleContext *aresample = ctx->priv; + int64_t out_rate, out_layout; + enum AVSampleFormat out_format; + char inchl_buf[128], outchl_buf[128]; + + aresample->swr = swr_alloc_set_opts(aresample->swr, + outlink->channel_layout, outlink->format, outlink->sample_rate, + inlink->channel_layout, inlink->format, inlink->sample_rate, + 0, ctx); + if (!aresample->swr) + return AVERROR(ENOMEM); + if (!inlink->channel_layout) + av_opt_set_int(aresample->swr, "ich", inlink->channels, 0); + if (!outlink->channel_layout) + av_opt_set_int(aresample->swr, "och", outlink->channels, 0); + + ret = swr_init(aresample->swr); + if (ret < 0) + return ret; + + av_opt_get_int(aresample->swr, "osr", 0, &out_rate); + av_opt_get_int(aresample->swr, "ocl", 0, &out_layout); + av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format); + outlink->time_base = (AVRational) {1, out_rate}; + + av_assert0(outlink->sample_rate == out_rate); + av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout); + av_assert0(outlink->format == out_format); + + aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate; + + av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout); + av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout); + + av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n", + inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate, + outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate); + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref) +{ + AResampleContext *aresample = inlink->dst->priv; + const int n_in = insamplesref->nb_samples; + int64_t delay; + int n_out = n_in * aresample->ratio + 32; + AVFilterLink *const outlink = inlink->dst->outputs[0]; + AVFrame *outsamplesref; + int ret; + + delay = swr_get_delay(aresample->swr, outlink->sample_rate); + if (delay > 0) + n_out += FFMIN(delay, FFMAX(4096, n_out)); + + outsamplesref = ff_get_audio_buffer(outlink, n_out); + + if(!outsamplesref) + return AVERROR(ENOMEM); + + av_frame_copy_props(outsamplesref, insamplesref); + outsamplesref->format = outlink->format; + av_frame_set_channels(outsamplesref, outlink->channels); + outsamplesref->channel_layout = outlink->channel_layout; + outsamplesref->sample_rate = outlink->sample_rate; + + if(insamplesref->pts != AV_NOPTS_VALUE) { + int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den); + int64_t outpts= swr_next_pts(aresample->swr, inpts); + aresample->next_pts = + outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate); + } else { + outsamplesref->pts = AV_NOPTS_VALUE; + } + n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, + (void *)insamplesref->extended_data, n_in); + if (n_out <= 0) { + av_frame_free(&outsamplesref); + av_frame_free(&insamplesref); + return 0; + } + + aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers + + outsamplesref->nb_samples = n_out; + + ret = ff_filter_frame(outlink, outsamplesref); + av_frame_free(&insamplesref); + return ret; +} + +static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret) +{ + AVFilterContext *ctx = outlink->src; + AResampleContext *aresample = ctx->priv; + AVFilterLink *const inlink = outlink->src->inputs[0]; + AVFrame *outsamplesref; + int n_out = 4096; + int64_t pts; + + outsamplesref = ff_get_audio_buffer(outlink, n_out); + *outsamplesref_ret = outsamplesref; + if (!outsamplesref) + return AVERROR(ENOMEM); + + pts = swr_next_pts(aresample->swr, INT64_MIN); + pts = ROUNDED_DIV(pts, inlink->sample_rate); + + n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0); + if (n_out <= 0) { + av_frame_free(&outsamplesref); + return (n_out == 0) ? AVERROR_EOF : n_out; + } + + outsamplesref->sample_rate = outlink->sample_rate; + outsamplesref->nb_samples = n_out; + + outsamplesref->pts = pts; + + return 0; +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AResampleContext *aresample = ctx->priv; + int ret; + + // First try to get data from the internal buffers + if (aresample->more_data) { + AVFrame *outsamplesref; + + if (flush_frame(outlink, 0, &outsamplesref) >= 0) { + return ff_filter_frame(outlink, outsamplesref); + } + } + aresample->more_data = 0; + + // Second request more data from the input + ret = ff_request_frame(ctx->inputs[0]); + + // Third if we hit the end flush + if (ret == AVERROR_EOF) { + AVFrame *outsamplesref; + + if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0) + return ret; + + return ff_filter_frame(outlink, outsamplesref); + } + return ret; +} + +static const AVClass *resample_child_class_next(const AVClass *prev) +{ + return prev ? NULL : swr_get_class(); +} + +static void *resample_child_next(void *obj, void *prev) +{ + AResampleContext *s = obj; + return prev ? NULL : s->swr; +} + +#define OFFSET(x) offsetof(AResampleContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption options[] = { + {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS }, + {NULL} +}; + +static const AVClass aresample_class = { + .class_name = "aresample", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, + .child_class_next = resample_child_class_next, + .child_next = resample_child_next, +}; + +static const AVFilterPad aresample_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad aresample_outputs[] = { + { + .name = "default", + .config_props = config_output, + .request_frame = request_frame, + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_aresample = { + .name = "aresample", + .description = NULL_IF_CONFIG_SMALL("Resample audio data."), + .init_dict = init_dict, + .uninit = uninit, + .query_formats = query_formats, + .priv_size = sizeof(AResampleContext), + .priv_class = &aresample_class, + .inputs = aresample_inputs, + .outputs = aresample_outputs, +}; |