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https://github.com/librempeg/librempeg
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03b078721c
* commit '97bf7c03b1338a867da52c159a2afecbdedcfa88': doc: git-howto: Leave reviewers time to react before pushing patches Include libavutil/channel_layout.h instead of libavutil/audioconvert.h lavu: rename audioconvert.* to channel_layout.* and deprecate audioconvert.h Conflicts: doc/APIchanges doc/examples/decoding_encoding.c doc/git-howto.texi ffmpeg_filter.c libavcodec/flacdec.c libavcodec/imc.c libavcodec/mpegaudiodec.c libavcodec/utils.c libavfilter/asrc_anullsrc.c libavfilter/audio.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/avfiltergraph.c libavfilter/buffer.c libavutil/Makefile libavutil/audioconvert.h libavutil/channel_layout.c libavutil/version.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
192 lines
5.4 KiB
C
192 lines
5.4 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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* buffer sink
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/mathematics.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "buffersink.h"
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#include "internal.h"
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typedef struct {
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AVFilterBufferRef *cur_buf; ///< last buffer delivered on the sink
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AVAudioFifo *audio_fifo; ///< FIFO for audio samples
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int64_t next_pts; ///< interpolating audio pts
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} BufferSinkContext;
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static av_cold void uninit(AVFilterContext *ctx)
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{
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BufferSinkContext *sink = ctx->priv;
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if (sink->audio_fifo)
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av_audio_fifo_free(sink->audio_fifo);
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}
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static int start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
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{
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BufferSinkContext *s = link->dst->priv;
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// av_assert0(!s->cur_buf);
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s->cur_buf = buf;
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link->cur_buf = NULL;
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return 0;
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}
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int ff_buffersink_read_compat(AVFilterContext *ctx, AVFilterBufferRef **buf)
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{
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BufferSinkContext *s = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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int ret;
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if (!buf)
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return ff_poll_frame(ctx->inputs[0]);
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if ((ret = ff_request_frame(link)) < 0)
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return ret;
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if (!s->cur_buf)
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return AVERROR(EINVAL);
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*buf = s->cur_buf;
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s->cur_buf = NULL;
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return 0;
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}
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static int read_from_fifo(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
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int nb_samples)
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{
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BufferSinkContext *s = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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AVFilterBufferRef *buf;
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if (!(buf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples)))
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return AVERROR(ENOMEM);
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av_audio_fifo_read(s->audio_fifo, (void**)buf->extended_data, nb_samples);
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buf->pts = s->next_pts;
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},
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link->time_base);
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*pbuf = buf;
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return 0;
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}
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int ff_buffersink_read_samples_compat(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
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int nb_samples)
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{
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BufferSinkContext *s = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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int ret = 0;
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if (!s->audio_fifo) {
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int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
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if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))
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return AVERROR(ENOMEM);
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}
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while (ret >= 0) {
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AVFilterBufferRef *buf;
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if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)
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return read_from_fifo(ctx, pbuf, nb_samples);
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ret = av_buffersink_read(ctx, &buf);
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if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo))
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return read_from_fifo(ctx, pbuf, av_audio_fifo_size(s->audio_fifo));
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else if (ret < 0)
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return ret;
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if (buf->pts != AV_NOPTS_VALUE) {
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s->next_pts = buf->pts -
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av_rescale_q(av_audio_fifo_size(s->audio_fifo),
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(AVRational){ 1, link->sample_rate },
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link->time_base);
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}
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ret = av_audio_fifo_write(s->audio_fifo, (void**)buf->extended_data,
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buf->audio->nb_samples);
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avfilter_unref_buffer(buf);
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}
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return ret;
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}
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static const AVFilterPad avfilter_vsink_buffer_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_VIDEO,
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.start_frame = start_frame,
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.min_perms = AV_PERM_READ,
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.needs_fifo = 1
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},
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{ NULL }
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};
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AVFilter avfilter_vsink_buffer = {
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#if AV_HAVE_INCOMPATIBLE_FORK_ABI
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.name = "buffersink",
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#else
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.name = "buffersink_old",
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#endif
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.description = NULL_IF_CONFIG_SMALL("Buffer video frames, and make them available to the end of the filter graph."),
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.priv_size = sizeof(BufferSinkContext),
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.uninit = uninit,
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.inputs = avfilter_vsink_buffer_inputs,
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.outputs = NULL,
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};
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static const AVFilterPad avfilter_asink_abuffer_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = start_frame,
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.min_perms = AV_PERM_READ,
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.needs_fifo = 1
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},
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{ NULL }
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};
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AVFilter avfilter_asink_abuffer = {
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#if AV_HAVE_INCOMPATIBLE_FORK_ABI
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.name = "abuffersink",
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#else
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.name = "abuffersink_old",
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#endif
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.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them available to the end of the filter graph."),
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.priv_size = sizeof(BufferSinkContext),
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.uninit = uninit,
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.inputs = avfilter_asink_abuffer_inputs,
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.outputs = NULL,
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};
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