diff options
Diffstat (limited to 'libavfilter/af_aresample.c')
-rw-r--r-- | libavfilter/af_aresample.c | 164 |
1 files changed, 164 insertions, 0 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c new file mode 100644 index 0000000000..521ccf7f8d --- /dev/null +++ b/libavfilter/af_aresample.c @@ -0,0 +1,164 @@ +/* + * 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 "libswresample/swresample.h" +#include "avfilter.h" +#include "audio.h" +#include "internal.h" + +typedef struct { + int out_rate; + double ratio; + struct SwrContext *swr; +} AResampleContext; + +static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque) +{ + AResampleContext *aresample = ctx->priv; + int ret; + + if (args) { + if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0) + return ret; + } else { + aresample->out_rate = -1; + } + + return 0; +} + +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; + + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + + AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO); + AVFilterFormats *out_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO); + AVFilterFormats *in_samplerates = ff_all_samplerates(); + AVFilterFormats *out_samplerates; + + + AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts(); + AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts(); + + avfilter_formats_ref(in_formats, &inlink->out_formats); + avfilter_formats_ref(out_formats, &outlink->in_formats); + + avfilter_formats_ref(in_samplerates, &inlink->out_samplerates); + + ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts); + ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts); + + if(aresample->out_rate > 0) { + int sample_rates[] = { aresample->out_rate, -1 }; + ff_set_common_samplerates(ctx, avfilter_make_format_list(sample_rates)); + } else { + out_samplerates = ff_all_samplerates(); + avfilter_formats_ref(out_samplerates, &outlink->in_samplerates); + } + + return 0; +} + + +static int config_output(AVFilterLink *outlink) +{ + int ret; + AVFilterContext *ctx = outlink->src; + AVFilterLink *inlink = ctx->inputs[0]; + AResampleContext *aresample = ctx->priv; + + if (aresample->out_rate == -1) + aresample->out_rate = outlink->sample_rate; + else + outlink->sample_rate = aresample->out_rate; + outlink->time_base = (AVRational) {1, aresample->out_rate}; + + //TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable + aresample->swr = swr_alloc_set_opts(aresample->swr, + inlink->channel_layout, inlink->format, aresample->out_rate, + inlink->channel_layout, inlink->format, inlink->sample_rate, + 0, ctx); + if (!aresample->swr) + return AVERROR(ENOMEM); + ret = swr_init(aresample->swr); + if (ret < 0) + return ret; + + aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate; + + av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n", + inlink->sample_rate, outlink->sample_rate); + return 0; +} + +static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) +{ + AResampleContext *aresample = inlink->dst->priv; + const int n_in = insamplesref->audio->nb_samples; + int n_out = n_in * aresample->ratio; + AVFilterLink *const outlink = inlink->dst->outputs[0]; + AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out); + + n_out = swr_convert(aresample->swr, outsamplesref->data, n_out, + (void *)insamplesref->data, n_in); + + avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); + outsamplesref->audio->sample_rate = outlink->sample_rate; + outsamplesref->audio->nb_samples = n_out; + outsamplesref->pts = insamplesref->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : + av_rescale(outlink->sample_rate, insamplesref->pts, inlink ->sample_rate); + + ff_filter_samples(outlink, outsamplesref); + avfilter_unref_buffer(insamplesref); +} + +AVFilter avfilter_af_aresample = { + .name = "aresample", + .description = NULL_IF_CONFIG_SMALL("Resample audio data."), + .init = init, + .uninit = uninit, + .query_formats = query_formats, + .priv_size = sizeof(AResampleContext), + + .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", + .config_props = config_output, + .type = AVMEDIA_TYPE_AUDIO, }, + { .name = NULL}}, +}; |