diff options
Diffstat (limited to 'libavfilter/af_aresample.c')
| -rw-r--r-- | libavfilter/af_aresample.c | 290 |
1 files changed, 290 insertions, 0 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c new file mode 100644 index 0000000..2e3867e --- /dev/null +++ b/libavfilter/af_aresample.c @@ -0,0 +1,290 @@ +/* + * 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 { + double ratio; + struct SwrContext *swr; + int64_t next_pts; + int req_fullfilled; +} AResampleContext; + +static av_cold int init(AVFilterContext *ctx, const char *args) +{ + AResampleContext *aresample = ctx->priv; + int ret = 0; + char *argd = av_strdup(args); + + aresample->next_pts = AV_NOPTS_VALUE; + aresample->swr = swr_alloc(); + if (!aresample->swr) { + ret = AVERROR(ENOMEM); + goto end; + } + + if (args) { + char *ptr = argd, *token; + + while (token = av_strtok(ptr, ":", &ptr)) { + char *value; + av_strtok(token, "=", &value); + + if (value) { + if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0) + goto end; + } else { + int out_rate; + if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0) + goto end; + if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0) + goto end; + } + } + } +end: + av_free(argd); + 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; + int out_rate = av_get_int(aresample->swr, "osr", NULL); + uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL); + enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL); + + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + + AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); + AVFilterFormats *out_formats; + AVFilterFormats *in_samplerates = ff_all_samplerates(); + AVFilterFormats *out_samplerates; + AVFilterChannelLayouts *in_layouts = ff_all_channel_counts(); + AVFilterChannelLayouts *out_layouts; + + ff_formats_ref (in_formats, &inlink->out_formats); + ff_formats_ref (in_samplerates, &inlink->out_samplerates); + ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts); + + if(out_rate > 0) { + out_samplerates = ff_make_format_list((int[]){ out_rate, -1 }); + } else { + out_samplerates = ff_all_samplerates(); + } + ff_formats_ref(out_samplerates, &outlink->in_samplerates); + + if(out_format != AV_SAMPLE_FMT_NONE) { + out_formats = ff_make_format_list((int[]){ out_format, -1 }); + } else + out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO); + ff_formats_ref(out_formats, &outlink->in_formats); + + if(out_layout) { + out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 }); + } else + out_layouts = ff_all_channel_counts(); + ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts); + + return 0; +} + + +static int config_output(AVFilterLink *outlink) +{ + int ret; + AVFilterContext *ctx = outlink->src; + AVFilterLink *inlink = ctx->inputs[0]; + AResampleContext *aresample = ctx->priv; + int out_rate; + uint64_t 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; + + out_rate = av_get_int(aresample->swr, "osr", NULL); + out_layout = av_get_int(aresample->swr, "ocl", NULL); + out_format = av_get_int(aresample->swr, "osf", NULL); + outlink->time_base = (AVRational) {1, out_rate}; + + av_assert0(outlink->sample_rate == out_rate); + av_assert0(outlink->channel_layout == out_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), -1, inlink ->channel_layout); + av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, 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, AVFilterBufferRef *insamplesref) +{ + AResampleContext *aresample = inlink->dst->priv; + const int n_in = insamplesref->audio->nb_samples; + int n_out = n_in * aresample->ratio * 2 + 256; + AVFilterLink *const outlink = inlink->dst->outputs[0]; + AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out); + int ret; + + if(!outsamplesref) + return AVERROR(ENOMEM); + + avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); + outsamplesref->format = outlink->format; + outsamplesref->audio->channels = outlink->channels; + outsamplesref->audio->channel_layout = outlink->channel_layout; + outsamplesref->audio->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) { + avfilter_unref_buffer(outsamplesref); + avfilter_unref_buffer(insamplesref); + return 0; + } + + outsamplesref->audio->nb_samples = n_out; + + ret = ff_filter_frame(outlink, outsamplesref); + aresample->req_fullfilled= 1; + avfilter_unref_buffer(insamplesref); + return ret; +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AResampleContext *aresample = ctx->priv; + AVFilterLink *const inlink = outlink->src->inputs[0]; + int ret; + + aresample->req_fullfilled = 0; + do{ + ret = ff_request_frame(ctx->inputs[0]); + }while(!aresample->req_fullfilled && ret>=0); + + if (ret == AVERROR_EOF) { + AVFilterBufferRef *outsamplesref; + int n_out = 4096; + + outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out); + if (!outsamplesref) + return AVERROR(ENOMEM); + n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0); + if (n_out <= 0) { + avfilter_unref_buffer(outsamplesref); + return (n_out == 0) ? AVERROR_EOF : n_out; + } + + outsamplesref->audio->sample_rate = outlink->sample_rate; + outsamplesref->audio->nb_samples = n_out; +#if 0 + outsamplesref->pts = aresample->next_pts; + if(aresample->next_pts != AV_NOPTS_VALUE) + aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base); +#else + outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN); + outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate); +#endif + + ff_filter_frame(outlink, outsamplesref); + return 0; + } + return ret; +} + +static const AVFilterPad aresample_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .min_perms = AV_PERM_READ, + }, + { NULL }, +}; + +static const AVFilterPad aresample_outputs[] = { + { + .name = "default", + .config_props = config_output, + .request_frame = request_frame, + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL }, +}; + +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 = aresample_inputs, + .outputs = aresample_outputs, +}; |
