e4e02a7d47
With this change avconv compiled against libav and linked to ffmpegs libs will run through the whole fate testsuite without any crashes. 857 tests pass, the remaining tests fail one way or another, which is to be expected as avconv is not a drop in replacement for ffmpeg The testsuite used was the ffmpeg fate testsuite, not libavs. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
180 lines
5.6 KiB
C
180 lines
5.6 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/audioconvert.h"
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#include "libavutil/avassert.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 av_buffersink_read(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 av_buffersink_read_samples(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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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 = (const AVFilterPad[]) {{ .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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{ .name = NULL }},
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.outputs = (const AVFilterPad[]) {{ .name = 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 = (const AVFilterPad[]) {{ .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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{ .name = NULL }},
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.outputs = (const AVFilterPad[]) {{ .name = NULL }},
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};
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