Add FLAC export and related options

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)

include/AudioFileFlac.h

@@ -0,0 +1,48 @@
+/*
+ * AudioFileFlac.h - Audio device which encodes a wave stream into a FLAC file.
+ */
+
+#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",

src/core/audio/AudioFileFlac.cpp

@@ -0,0 +1,95 @@
+#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

@@ -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"