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https://github.com/FFmpeg/FFmpeg.git
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f8911b987d
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
147 lines
4.5 KiB
C
147 lines
4.5 KiB
C
/*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Channel split filter
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*
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* Split an audio stream into per-channel streams.
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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typedef struct ChannelSplitContext {
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const AVClass *class;
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uint64_t channel_layout;
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char *channel_layout_str;
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} ChannelSplitContext;
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#define OFFSET(x) offsetof(ChannelSplitContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption channelsplit_options[] = {
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{ "channel_layout", "Input channel layout.", OFFSET(channel_layout_str), AV_OPT_TYPE_STRING, { .str = "stereo" }, .flags = A },
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{ NULL },
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};
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AVFILTER_DEFINE_CLASS(channelsplit);
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static int init(AVFilterContext *ctx, const char *arg)
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{
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ChannelSplitContext *s = ctx->priv;
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int nb_channels;
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int ret = 0, i;
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s->class = &channelsplit_class;
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av_opt_set_defaults(s);
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if ((ret = av_set_options_string(s, arg, "=", ":")) < 0) {
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av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", arg);
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return ret;
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}
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if (!(s->channel_layout = av_get_channel_layout(s->channel_layout_str))) {
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av_log(ctx, AV_LOG_ERROR, "Error parsing channel layout '%s'.\n",
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s->channel_layout_str);
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ret = AVERROR(EINVAL);
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goto fail;
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}
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nb_channels = av_get_channel_layout_nb_channels(s->channel_layout);
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for (i = 0; i < nb_channels; i++) {
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uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
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AVFilterPad pad = { 0 };
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pad.type = AVMEDIA_TYPE_AUDIO;
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pad.name = av_get_channel_name(channel);
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ff_insert_outpad(ctx, i, &pad);
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}
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fail:
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av_opt_free(s);
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return ret;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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ChannelSplitContext *s = ctx->priv;
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AVFilterChannelLayouts *in_layouts = NULL;
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int i;
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ff_set_common_formats (ctx, ff_planar_sample_fmts());
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ff_set_common_samplerates(ctx, ff_all_samplerates());
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ff_add_channel_layout(&in_layouts, s->channel_layout);
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ff_channel_layouts_ref(in_layouts, &ctx->inputs[0]->out_channel_layouts);
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for (i = 0; i < ctx->nb_outputs; i++) {
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AVFilterChannelLayouts *out_layouts = NULL;
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uint64_t channel = av_channel_layout_extract_channel(s->channel_layout, i);
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ff_add_channel_layout(&out_layouts, channel);
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ff_channel_layouts_ref(out_layouts, &ctx->outputs[i]->in_channel_layouts);
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}
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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int i, ret = 0;
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for (i = 0; i < ctx->nb_outputs; i++) {
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AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
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if (!buf_out) {
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ret = AVERROR(ENOMEM);
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break;
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}
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buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
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buf_out->audio->channel_layout =
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av_channel_layout_extract_channel(buf->audio->channel_layout, i);
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ret = ff_filter_samples(ctx->outputs[i], buf_out);
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if (ret < 0)
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break;
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}
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avfilter_unref_buffer(buf);
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return ret;
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}
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AVFilter avfilter_af_channelsplit = {
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.name = "channelsplit",
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.description = NULL_IF_CONFIG_SMALL("Split audio into per-channel streams"),
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.priv_size = sizeof(ChannelSplitContext),
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.init = init,
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.query_formats = query_formats,
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.inputs = (const AVFilterPad[]){{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples, },
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{ NULL }},
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.outputs = (const AVFilterPad[]){{ NULL }},
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};
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