filter/hdcd: move generic code to class FfmpegFilter
This commit is contained in:
Max Kellermann
parent
9f62824e98
commit
c0d6008781
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@ -0,0 +1,76 @@
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/*
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* Copyright 2003-2019 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "FfmpegFilter.hxx"
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#include "lib/ffmpeg/SampleFormat.hxx"
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#include "util/ConstBuffer.hxx"
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extern "C" {
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#include <libavfilter/buffersrc.h>
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#include <libavfilter/buffersink.h>
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}
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FfmpegFilter::FfmpegFilter(const AudioFormat &in_audio_format,
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const AudioFormat &_out_audio_format,
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Ffmpeg::FilterGraph &&_graph,
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Ffmpeg::FilterContext &&_buffer_src,
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Ffmpeg::FilterContext &&_buffer_sink) noexcept
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:Filter(_out_audio_format),
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graph(std::move(_graph)),
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buffer_src(std::move(_buffer_src)),
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buffer_sink(std::move(_buffer_sink)),
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in_audio_frame_size(in_audio_format.GetFrameSize()),
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out_audio_frame_size(_out_audio_format.GetFrameSize())
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{
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in_frame->format = Ffmpeg::ToFfmpegSampleFormat(in_audio_format.format);
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in_frame->sample_rate = in_audio_format.sample_rate;
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in_frame->channels = in_audio_format.channels;
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}
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ConstBuffer<void>
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FfmpegFilter::FilterPCM(ConstBuffer<void> src)
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{
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/* submit source data into the FFmpeg audio buffer source */
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in_frame->nb_samples = src.size / in_audio_frame_size;
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in_frame.GetBuffer();
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in_frame.MakeWritable();
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memcpy(in_frame.GetData(0), src.data, src.size);
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int err = av_buffersrc_write_frame(buffer_src.get(), in_frame.get());
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if (err < 0)
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throw MakeFfmpegError(err, "av_buffersrc_write_frame() failed");
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/* collect filtered data from the FFmpeg audio buffer sink */
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err = av_buffersink_get_frame(buffer_sink.get(), out_frame.get());
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if (err < 0) {
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if (err == AVERROR(EAGAIN) || err == AVERROR_EOF)
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return nullptr;
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throw MakeFfmpegError(err, "av_buffersink_get_frame() failed");
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}
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/* TODO: call av_buffersink_get_frame() repeatedly? Not
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possible with MPD's current Filter API */
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return {out_frame.GetData(0), out_frame->nb_samples * GetOutAudioFormat().GetFrameSize()};
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}
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@ -0,0 +1,56 @@
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/*
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* Copyright 2003-2019 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef MPD_FFMPEG_FILTER__HXX
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#define MPD_FFMPEG_FILTER__HXX
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#include "filter/Filter.hxx"
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#include "lib/ffmpeg/Filter.hxx"
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#include "lib/ffmpeg/Frame.hxx"
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/**
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* A #Filter implementation using FFmpeg's libavfilter.
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*/
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class FfmpegFilter final : public Filter {
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Ffmpeg::FilterGraph graph;
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Ffmpeg::FilterContext buffer_src, buffer_sink;
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Ffmpeg::Frame in_frame, out_frame;
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const size_t in_audio_frame_size;
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const size_t out_audio_frame_size;
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public:
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/**
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* @param _graph a checked and configured AVFilterGraph
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* @param _buffer_src an "abuffer" filter which serves as
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* input
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* @param _buffer_sink an "abuffersink" filter which serves as
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* output
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*/
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FfmpegFilter(const AudioFormat &in_audio_format,
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const AudioFormat &_out_audio_format,
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Ffmpeg::FilterGraph &&_graph,
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Ffmpeg::FilterContext &&_buffer_src,
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Ffmpeg::FilterContext &&_buffer_sink) noexcept;
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/* virtual methods from class Filter */
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ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
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};
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#endif
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@ -18,6 +18,7 @@
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*/
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#include "HdcdFilterPlugin.hxx"
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#include "FfmpegFilter.hxx"
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#include "filter/FilterPlugin.hxx"
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#include "filter/Filter.hxx"
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#include "filter/NullFilter.hxx"
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@ -27,12 +28,6 @@
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#include "lib/ffmpeg/SampleFormat.hxx"
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#include "config/Block.hxx"
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#include "AudioFormat.hxx"
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#include "util/ConstBuffer.hxx"
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extern "C" {
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#include <libavfilter/buffersrc.h>
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#include <libavfilter/buffersink.h>
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}
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#include <string.h>
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@ -47,32 +42,20 @@ MaybeHdcd(const AudioFormat &audio_format) noexcept
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audio_format.channels == 2;
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}
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class HdcdFilter final : public Filter {
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Ffmpeg::FilterGraph graph;
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Ffmpeg::FilterContext buffer_src, buffer_sink;
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Ffmpeg::Frame frame, out_frame;
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size_t in_audio_frame_size;
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size_t out_audio_frame_size;
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public:
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explicit HdcdFilter(AudioFormat &audio_format);
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/* virtual methods from class Filter */
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ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
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};
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inline
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HdcdFilter::HdcdFilter(AudioFormat &audio_format)
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:Filter(audio_format)
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static auto
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OpenHdcdFilter(AudioFormat &in_audio_format)
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{
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buffer_src = Ffmpeg::FilterContext::MakeAudioBufferSource(audio_format,
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*graph);
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Ffmpeg::FilterGraph graph;
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buffer_sink = Ffmpeg::FilterContext::MakeAudioBufferSink(*graph);
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auto buffer_src =
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Ffmpeg::FilterContext::MakeAudioBufferSource(in_audio_format,
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*graph);
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auto buffer_sink = Ffmpeg::FilterContext::MakeAudioBufferSink(*graph);
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Ffmpeg::FilterInOut io_sink("out", *buffer_sink);
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Ffmpeg::FilterInOut io_src("in", *buffer_src);
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auto io = graph.Parse(hdcd_graph, std::move(io_sink),
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std::move(io_src));
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@ -84,15 +67,15 @@ HdcdFilter::HdcdFilter(AudioFormat &audio_format)
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graph.CheckAndConfigure();
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frame->format = Ffmpeg::ToFfmpegSampleFormat(audio_format.format);
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frame->sample_rate = audio_format.sample_rate;
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frame->channels = audio_format.channels;
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auto out_audio_format = in_audio_format;
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// TODO: convert to 32 bit only if HDCD actually detected
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out_audio_format.format = SampleFormat::S32;
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in_audio_frame_size = audio_format.GetFrameSize();
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out_audio_frame_size = out_audio_format.GetFrameSize();
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return std::make_unique<FfmpegFilter>(in_audio_format,
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out_audio_format,
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std::move(graph),
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std::move(buffer_src),
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std::move(buffer_sink));
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}
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class PreparedHdcdFilter final : public PreparedFilter {
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@ -105,45 +88,13 @@ std::unique_ptr<Filter>
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PreparedHdcdFilter::Open(AudioFormat &audio_format)
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{
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if (MaybeHdcd(audio_format))
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return std::make_unique<HdcdFilter>(audio_format);
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return OpenHdcdFilter(audio_format);
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else
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/* this cannot be HDCD, so let's copy as-is using
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NullFilter */
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return std::make_unique<NullFilter>(audio_format);
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}
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ConstBuffer<void>
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HdcdFilter::FilterPCM(ConstBuffer<void> src)
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{
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/* submit source data into the FFmpeg audio buffer source */
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frame->nb_samples = src.size / in_audio_frame_size;
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frame.GetBuffer();
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frame.MakeWritable();
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memcpy(frame.GetData(0), src.data, src.size);
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int err = av_buffersrc_write_frame(buffer_src.get(), frame.get());
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if (err < 0)
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throw MakeFfmpegError(err, "av_buffersrc_write_frame() failed");
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/* collect filtered data from the FFmpeg audio buffer sink */
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err = av_buffersink_get_frame(buffer_sink.get(), out_frame.get());
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if (err < 0) {
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if (err == AVERROR(EAGAIN) || err == AVERROR_EOF)
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return nullptr;
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throw MakeFfmpegError(err, "av_buffersink_get_frame() failed");
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}
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/* TODO: call av_buffersink_get_frame() repeatedly? Not
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possible with MPD's current Filter API */
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return {out_frame.GetData(0), out_frame->nb_samples * GetOutAudioFormat().GetFrameSize()};
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}
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static std::unique_ptr<PreparedFilter>
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hdcd_filter_init(const ConfigBlock &)
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{
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@ -3,6 +3,7 @@ filter_plugins_deps = []
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if libavfilter_dep.found()
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filter_plugins_sources += [
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'FfmpegFilter.cxx',
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'HdcdFilterPlugin.cxx',
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]
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filter_plugins_deps += ffmpeg_dep
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