Add FLAC export and related options (LMMS#3731)

* Add FLAC export, based on WAV renderer

* Depend on sndfile>=1.0.18 (previously >=1.0.11)

* Add compression option to FLAC export, if available.

The code related to compression is only generated if LMMS_HAVE_SF_COMPLEVEL is defined(libsndfile>=1.0.26).

* Save into the correct file extension upon single-export.

* Use unique_ptr in FLAC renderer and be more expressive about involved types.

* Use unique_ptr and remove manual memory management in ExportProjectDialog

* Add 'flac' format info to --help and manpage

CMakeLists.txt

@@ -176,10 +176,14 @@ ELSE()
 ENDIF()
 
 # check for libsndfile
-PKG_CHECK_MODULES(SNDFILE REQUIRED sndfile>=1.0.11)
+PKG_CHECK_MODULES(SNDFILE REQUIRED sndfile>=1.0.18)
 IF(NOT SNDFILE_FOUND)
-	MESSAGE(FATAL_ERROR "LMMS requires libsndfile1 and libsndfile1-dev >= 1.0.11 - please install, remove CMakeCache.txt and try again!")
-ENDIF(NOT SNDFILE_FOUND)
+	MESSAGE(FATAL_ERROR "LMMS requires libsndfile1 and libsndfile1-dev >= 1.0.18 - please install, remove CMakeCache.txt and try again!")
+ENDIF()
+# check if we can use SF_SET_COMPRESSION_LEVEL
+IF(NOT SNDFILE_VERSION VERSION_LESS 1.0.26)
+	SET(LMMS_HAVE_SF_COMPLEVEL TRUE)
+ENDIF()
 
 IF(WANT_CALF)
 	SET(LMMS_HAVE_CALF TRUE)
@@ -625,6 +629,7 @@ MESSAGE(
 "Supported file formats for project export\n"
 "-----------------------------------------\n"
 "* WAVE                        : OK\n"
+"* FLAC                        : OK\n"
 "* OGG/VORBIS                  : ${STATUS_OGGVORBIS}\n"
 "* MP3/Lame                    : ${STATUS_MP3LAME}\n"
 )

doc/lmms.1

@@ -97,7 +97,7 @@ Get the configuration from \fIconfigfile\fP instead of ~/.lmmsrc.xml (default)
 .IP "\fB\-d, --dump\fP \fIin\fP
 Dump XML of compressed file \fIin\fP (i.e. MMPZ-file)
 .IP "\fB\-f, --format\fP \fIformat\fP
-Specify format of render-output where \fIformat\fP is either 'wav', 'ogg' or 'mp3'.
+Specify format of render-output where \fIformat\fP is either 'wav', 'flac', 'ogg' or 'mp3'.
 .IP "\fB\    --geometry\fP \fIgeometry\fP
 Specify the prefered size and position of the main window
 .br

include/AudioFileFlac.h

@@ -0,0 +1,74 @@
+/*
+ * AudioFileFlac.h - Audio device which encodes a wave stream into a FLAC file.
+ *
+ * Copyright (c) 2017 to present Levin Oehlmann <irrenhaus3/at/gmail[dot]com> et al.
+ *
+ * This file is part of LMMS - https://lmms.io
+ *
+ * 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 (see COPYING); if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ * Boston, MA 02110-1301 USA.
+ *
+ */
+
+#ifndef AUDIO_FILE_FLAC_H
+#define AUDIO_FILE_FLAC_H
+
+#include "lmmsconfig.h"
+
+#include "AudioFileDevice.h"
+#include <sndfile.h>
+
+class AudioFileFlac: public AudioFileDevice
+{
+public:
+	AudioFileFlac(OutputSettings const& outputSettings,
+			ch_cnt_t const channels,
+			bool& successful,
+			QString const& file,
+			Mixer* mixer
+	);
+
+	virtual ~AudioFileFlac();
+
+	static AudioFileDevice* getInst(QString const& outputFilename,
+			OutputSettings const& outputSettings,
+			ch_cnt_t const channels,
+			Mixer* mixer,
+			bool& successful)
+	{
+		return new AudioFileFlac(
+			outputSettings,
+			channels,
+			successful,
+			outputFilename,
+			mixer
+		);
+	}
+
+private:
+
+	SF_INFO  m_sfinfo;
+	SNDFILE* m_sf;
+
+	virtual void writeBuffer(surroundSampleFrame const* _ab,
+						fpp_t const frames,
+						float master_gain) override;
+
+	bool startEncoding();
+	void finishEncoding();
+
+};
+
+#endif //AUDIO_FILE_FLAC_H

include/ExportProjectDialog.h

@@ -28,7 +28,7 @@
 #define EXPORT_PROJECT_DIALOG_H
 
 #include <QDialog>
-#include <vector>
+#include <memory>
 #include "ui_export_project.h"
 
 #include "ProjectRenderer.h"
@@ -39,8 +39,6 @@ class ExportProjectDialog : public QDialog, public Ui::ExportProjectDialog
 	Q_OBJECT
 public:
 	ExportProjectDialog( const QString & _file_name, QWidget * _parent, bool multi_export );
-	virtual ~ExportProjectDialog();
-
 
 protected:
 	virtual void reject( void );
@@ -62,7 +60,7 @@ private slots:
 	bool m_multiExport;
 
 	ProjectRenderer::ExportFileFormats m_ft;
-	RenderManager* m_renderManager;
+	std::unique_ptr<RenderManager> m_renderManager;
 } ;
 
 #endif

include/OutputSettings.h

@@ -72,7 +72,8 @@ class OutputSettings
 		m_sampleRate(sampleRate),
 		m_bitRateSettings(bitRateSettings),
 		m_bitDepth(bitDepth),
-		m_stereoMode(stereoMode)
+		m_stereoMode(stereoMode),
+		m_compressionLevel(0.5)
 	{
 	}
 
@@ -95,11 +96,19 @@ class OutputSettings
 	StereoMode getStereoMode() const { return m_stereoMode; }
 	void setStereoMode(StereoMode stereoMode) { m_stereoMode = stereoMode; }
 
+
+	double getCompressionLevel() const{ return m_compressionLevel; }
+	void setCompressionLevel(double level){
+		// legal range is 0.0 to 1.0.
+		m_compressionLevel = level;
+	}
+
 private:
 	sample_rate_t m_sampleRate;
 	BitRateSettings m_bitRateSettings;
 	BitDepth m_bitDepth;
 	StereoMode m_stereoMode;
+	double m_compressionLevel;
 };
 
 #endif

include/ProjectRenderer.h

@@ -38,6 +38,7 @@ class ProjectRenderer : public QThread
 	enum ExportFileFormats: int
 	{
 		WaveFile,
+		FlacFile,
 		OggFile,
 		MP3File,
 		NumFileFormats

src/core/CMakeLists.txt

@@ -71,6 +71,7 @@ set(LMMS_SRCS
 	core/audio/AudioFileDevice.cpp
 	core/audio/AudioFileMP3.cpp
 	core/audio/AudioFileOgg.cpp
+	core/audio/AudioFileFlac.cpp
 	core/audio/AudioFileWave.cpp
 	core/audio/AudioJack.cpp
 	core/audio/AudioOss.cpp

src/core/ProjectRenderer.cpp

@@ -31,6 +31,7 @@
 #include "AudioFileWave.h"
 #include "AudioFileOgg.h"
 #include "AudioFileMP3.h"
+#include "AudioFileFlac.h"
 
 #ifdef LMMS_HAVE_SCHED_H
 #include "sched.h"
@@ -42,6 +43,11 @@ const ProjectRenderer::FileEncodeDevice ProjectRenderer::fileEncodeDevices[] =
 	{ ProjectRenderer::WaveFile,
 		QT_TRANSLATE_NOOP( "ProjectRenderer", "WAV-File (*.wav)" ),
 					".wav", &AudioFileWave::getInst },
+	{ ProjectRenderer::FlacFile,
+		QT_TRANSLATE_NOOP("ProjectRenderer", "FLAC-File (*.flac)"),
+		".flac",
+		&AudioFileFlac::getInst
+	},
 	{ ProjectRenderer::OggFile,
 		QT_TRANSLATE_NOOP( "ProjectRenderer", "Compressed OGG-File (*.ogg)" ),
 					".ogg",
@@ -176,8 +182,8 @@ void ProjectRenderer::run()
 
 	Engine::getSong()->startExport();
 	Engine::getSong()->updateLength();
-    //skip first empty buffer
-    Engine::mixer()->nextBuffer();
+	//skip first empty buffer
+	Engine::mixer()->nextBuffer();
 
 	const Song::PlayPos & exportPos = Engine::getSong()->getPlayPos(
 							Song::Mode_PlaySong );

src/core/audio/AudioFileFlac.cpp

@@ -0,0 +1,119 @@
+/*
+ * AudioFileFlac.cpp - Audio device which encodes a wave stream into a FLAC file (Implementation).
+ *
+ * Copyright (c) 2017 to present Levin Oehlmann <irrenhaus3/at/gmail[dot]com> et al.
+ *
+ * This file is part of LMMS - https://lmms.io
+ *
+ * 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 (see COPYING); if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ * Boston, MA 02110-1301 USA.
+ *
+ */
+
+#include <memory>
+
+#include "AudioFileFlac.h"
+#include "endian_handling.h"
+#include "Mixer.h"
+
+AudioFileFlac::AudioFileFlac(OutputSettings const& outputSettings, ch_cnt_t const channels, bool& successful, QString const& file, Mixer* mixer):
+	AudioFileDevice(outputSettings,channels,file,mixer),
+	m_sf(nullptr)
+{
+	successful = outputFileOpened() && startEncoding();
+}
+
+AudioFileFlac::~AudioFileFlac()
+{
+	finishEncoding();
+}
+
+bool AudioFileFlac::startEncoding()
+{
+	m_sfinfo.samplerate=sampleRate();
+	m_sfinfo.channels=channels();
+	m_sfinfo.frames = mixer()->framesPerPeriod();
+	m_sfinfo.sections=1;
+	m_sfinfo.seekable=0;
+
+	m_sfinfo.format = SF_FORMAT_FLAC;
+
+	switch (getOutputSettings().getBitDepth())
+	{
+		case OutputSettings::Depth_24Bit:
+		case OutputSettings::Depth_32Bit:
+			// FLAC does not support 32bit sampling, so take it as 24.
+			m_sfinfo.format |= SF_FORMAT_PCM_24;
+			break;
+		default:
+			m_sfinfo.format |= SF_FORMAT_PCM_16;
+	}
+
+#ifdef LMMS_HAVE_SF_COMPLEVEL
+	double compression = getOutputSettings().getCompressionLevel();
+	sf_command(m_sf, SFC_SET_COMPRESSION_LEVEL, &compression, sizeof(double));
+#endif
+
+	m_sf = sf_open(
+#ifdef LMMS_BUILD_WIN32
+		outputFile().toLocal8Bit().constData(),
+#else
+		outputFile().toUtf8().constData(),
+#endif
+		SFM_WRITE,
+		&m_sfinfo
+	);
+
+	sf_command(m_sf, SFC_SET_CLIPPING, nullptr, SF_TRUE);
+
+	sf_set_string(m_sf, SF_STR_SOFTWARE, "LMMS");
+
+	return true;
+}
+
+void AudioFileFlac::writeBuffer(surroundSampleFrame const* _ab, fpp_t const frames, float master_gain)
+{
+	OutputSettings::BitDepth depth = getOutputSettings().getBitDepth();
+
+	if (depth == OutputSettings::Depth_24Bit || depth == OutputSettings::Depth_32Bit) // Float encoding
+	{
+		std::unique_ptr<sample_t[]> buf{ new sample_t[frames*channels()] };
+		for(fpp_t frame = 0; frame < frames; ++frame)
+		{
+			for(ch_cnt_t channel=0; channel<channels(); ++channel)
+			{
+				buf[frame*channels() + channel] = _ab[frame][channel] * master_gain;
+			}
+		}
+		sf_writef_float(m_sf,static_cast<float*>(buf.get()),frames);
+	}
+	else // integer PCM encoding
+	{
+		std::unique_ptr<int_sample_t[]> buf{ new int_sample_t[frames*channels()] };
+		convertToS16(_ab, frames, master_gain, buf.get(), !isLittleEndian());
+		sf_writef_short(m_sf, static_cast<short*>(buf.get()), frames);
+	}
+
+}
+
+
+void AudioFileFlac::finishEncoding()
+{
+	if (m_sf)
+	{
+		sf_write_sync(m_sf);
+		sf_close(m_sf);
+	}
+}

src/core/main.cpp

@@ -166,7 +166,7 @@ void printHelp()
 		"-c, --config <configfile>     Get the configuration from <configfile>\n"
 		"-d, --dump <in>               Dump XML of compressed file <in>\n"
 		"-f, --format <format>         Specify format of render-output where\n"
-		"       Format is either 'wav', 'ogg' or 'mp3'.\n"
+		"       Format is either 'wav', 'flac', 'ogg' or 'mp3'.\n"
 		"    --geometry <geometry>     Specify the size and position of the main window\n"
 		"       geometry is <xsizexysize+xoffset+yoffsety>.\n"
 		"-h, --help                    Show this usage information and exit.\n"
@@ -444,6 +444,10 @@ int main( int argc, char * * argv )
 				eff = ProjectRenderer::MP3File;
 			}
 #endif
+			else if (ext == "flac")
+			{
+				eff = ProjectRenderer::FlacFile;
+			}
 			else
 			{
 				printf( "\nInvalid output format %s.\n\n"