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/*
* Copyright 2003-2020 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "FfmpegFilter.hxx"
#include "lib/ffmpeg/Interleave.hxx"
#include "lib/ffmpeg/SampleFormat.hxx"
#include "util/ConstBuffer.hxx"
extern "C" {
#include <libavfilter/buffersrc.h>
#include <libavfilter/buffersink.h>
}
#include <string.h>
FfmpegFilter::FfmpegFilter(const AudioFormat &in_audio_format,
const AudioFormat &_out_audio_format,
Ffmpeg::FilterGraph &&_graph,
Ffmpeg::FilterContext &&_buffer_src,
Ffmpeg::FilterContext &&_buffer_sink) noexcept
:Filter(_out_audio_format),
graph(std::move(_graph)),
buffer_src(std::move(_buffer_src)),
buffer_sink(std::move(_buffer_sink)),
in_format(Ffmpeg::ToFfmpegSampleFormat(in_audio_format.format)),
in_sample_rate(in_audio_format.sample_rate),
in_channels(in_audio_format.channels),
in_audio_frame_size(in_audio_format.GetFrameSize()),
out_audio_frame_size(_out_audio_format.GetFrameSize())
{
}
ConstBuffer<void>
FfmpegFilter::FilterPCM(ConstBuffer<void> src)
{
/* submit source data into the FFmpeg audio buffer source */
frame.Unref();
frame->format = in_format;
frame->sample_rate = in_sample_rate;
frame->channels = in_channels;
frame->nb_samples = src.size / in_audio_frame_size;
frame.GetBuffer();
memcpy(frame.GetData(0), src.data, src.size);
int err = av_buffersrc_add_frame(buffer_src.get(), frame.get());
if (err < 0)
throw MakeFfmpegError(err, "av_buffersrc_write_frame() failed");
/* collect filtered data from the FFmpeg audio buffer sink */
frame.Unref();
err = av_buffersink_get_frame(buffer_sink.get(), frame.get());
if (err < 0) {
if (err == AVERROR(EAGAIN) || err == AVERROR_EOF)
return nullptr;
throw MakeFfmpegError(err, "av_buffersink_get_frame() failed");
}
/* TODO: call av_buffersink_get_frame() repeatedly? Not
possible with MPD's current Filter API */
return Ffmpeg::InterleaveFrame(*frame, interleave_buffer);
}
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