Apply master gain outside audio devices

include/AudioDevice.h

@@ -96,11 +96,7 @@ class AudioDevice
 protected:
 	// subclasses can re-implement this for being used in conjunction with
 	// processNextBuffer()
-	virtual void writeBuffer( const surroundSampleFrame * /* _buf*/,
-						const fpp_t /*_frames*/,
-						const float /*_master_gain*/ )
-	{
-	}
+	virtual void writeBuffer(const surroundSampleFrame* /* _buf*/, const fpp_t /*_frames*/) {}
 
 	// called by according driver for fetching new sound-data
 	fpp_t getNextBuffer( surroundSampleFrame * _ab );
@@ -109,7 +105,6 @@ class AudioDevice
 	// returns num of bytes in outbuf
 	int convertToS16( const surroundSampleFrame * _ab,
 						const fpp_t _frames,
-						const float _master_gain,
 						int_sample_t * _output_buffer,
 						const bool _convert_endian = false );
 

include/AudioFileFlac.h

@@ -65,9 +65,7 @@ class AudioFileFlac : public AudioFileDevice
 	SF_INFO  m_sfinfo;
 	SNDFILE* m_sf;
 
-	void writeBuffer(surroundSampleFrame const* _ab,
-						fpp_t const frames,
-						float master_gain) override;
+	void writeBuffer(surroundSampleFrame const* _ab, fpp_t const frames) override;
 
 	bool startEncoding();
 	void finishEncoding();

include/AudioFileMP3.h

@@ -58,9 +58,7 @@ class AudioFileMP3 : public AudioFileDevice
 	}
 
 protected:
-	void writeBuffer( const surroundSampleFrame * /* _buf*/,
-				  const fpp_t /*_frames*/,
-				  const float /*_master_gain*/ ) override;
+	void writeBuffer(const surroundSampleFrame* /* _buf*/, const fpp_t /*_frames*/) override;
 
 private:
 	void flushRemainingBuffers();

include/AudioFileOgg.h

@@ -58,9 +58,7 @@ class AudioFileOgg : public AudioFileDevice
 
 
 private:
-	void writeBuffer( const surroundSampleFrame * _ab,
-						const fpp_t _frames,
-						const float _master_gain ) override;
+	void writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames) override;
 
 	bool startEncoding();
 	void finishEncoding();

include/AudioFileWave.h

@@ -56,9 +56,7 @@ class AudioFileWave : public AudioFileDevice
 
 
 private:
-	void writeBuffer( const surroundSampleFrame * _ab,
-						const fpp_t _frames,
-						float _master_gain ) override;
+	void writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames) override;
 
 	bool startEncoding();
 	void finishEncoding();

include/AudioSampleRecorder.h

@@ -48,9 +48,7 @@ class AudioSampleRecorder : public AudioDevice
 	std::shared_ptr<const SampleBuffer> createSampleBuffer();
 
 private:
-	void writeBuffer( const surroundSampleFrame * _ab,
-						const fpp_t _frames,
-						const float _master_gain ) override;
+	void writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames) override;
 
 	using BufferList = QList<QPair<sampleFrame*, fpp_t>>;
 	BufferList m_buffers;

include/MixHelpers.h

@@ -45,6 +45,8 @@ bool sanitize( sampleFrame * src, int frames );
 /*! \brief Add samples from src to dst */
 void add( sampleFrame* dst, const sampleFrame* src, int frames );
 
+/*! \brief Multiply samples from `dst` by `coeff` */
+void multiply(sampleFrame* dst, float coeff, int frames);
 
 /*! \brief Add samples from src multiplied by coeffSrc to dst */
 void addMultiplied( sampleFrame* dst, const sampleFrame* src, float coeffSrc, int frames );

src/core/AudioEngine.cpp

@@ -24,6 +24,7 @@
 
 #include "AudioEngine.h"
 
+#include "MixHelpers.h"
 #include "denormals.h"
 
 #include "lmmsconfig.h"
@@ -433,6 +434,8 @@ void AudioEngine::renderStageMix()
 	Mixer *mixer = Engine::mixer();
 	mixer->masterMix(m_outputBufferWrite);
 
+	MixHelpers::multiply(m_outputBufferWrite, m_masterGain, m_framesPerPeriod);
+
 	emit nextAudioBuffer(m_outputBufferRead);
 
 	// and trigger LFOs

src/core/MixHelpers.cpp

@@ -178,6 +178,15 @@ struct AddSwappedMultipliedOp
 	const float m_coeff;
 };
 
+void multiply(sampleFrame* dst, float coeff, int frames)
+{
+	for (int i = 0; i < frames; ++i)
+	{
+		dst[i][0] *= coeff;
+		dst[i][1] *= coeff;
+	}
+}
+
 void addSwappedMultiplied( sampleFrame* dst, const sampleFrame* src, float coeffSrc, int frames )
 {
 	run<>( dst, src, frames, AddSwappedMultipliedOp(coeffSrc) );

src/core/audio/AudioAlsa.cpp

@@ -323,10 +323,7 @@ void AudioAlsa::run()
 				}
 				outbuf_size = frames * channels();
 
-				convertToS16( temp, frames,
-						audioEngine()->masterGain(),
-						outbuf,
-						m_convertEndian );
+				convertToS16(temp, frames, outbuf, m_convertEndian);
 			}
 			int min_len = std::min(len, outbuf_size - outbuf_pos);
 			memcpy( ptr, outbuf + outbuf_pos,

src/core/audio/AudioDevice.cpp

@@ -66,10 +66,7 @@ AudioDevice::~AudioDevice()
 void AudioDevice::processNextBuffer()
 {
 	const fpp_t frames = getNextBuffer( m_buffer );
-	if( frames )
-	{
-		writeBuffer( m_buffer, frames, audioEngine()->masterGain() );
-	}
+	if (frames) { writeBuffer(m_buffer, frames); }
 	else
 	{
 		m_inProcess = false;
@@ -211,7 +208,6 @@ fpp_t AudioDevice::resample( const surroundSampleFrame * _src,
 
 int AudioDevice::convertToS16( const surroundSampleFrame * _ab,
 								const fpp_t _frames,
-								const float _master_gain,
 								int_sample_t * _output_buffer,
 								const bool _convert_endian )
 {
@@ -222,8 +218,8 @@ int AudioDevice::convertToS16( const surroundSampleFrame * _ab,
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
-				temp = static_cast<int_sample_t>( AudioEngine::clip( _ab[frame][chnl] * _master_gain ) * OUTPUT_SAMPLE_MULTIPLIER );
-				
+				temp = static_cast<int_sample_t>(AudioEngine::clip(_ab[frame][chnl]) * OUTPUT_SAMPLE_MULTIPLIER);
+
 				( _output_buffer + frame * channels() )[chnl] =
 						( temp & 0x00ff ) << 8 |
 						( temp & 0xff00 ) >> 8;
@@ -236,11 +232,8 @@ int AudioDevice::convertToS16( const surroundSampleFrame * _ab,
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
-				( _output_buffer + frame * channels() )[chnl] =
-						static_cast<int_sample_t>(
-						AudioEngine::clip( _ab[frame][chnl] *
-						_master_gain ) *
-						OUTPUT_SAMPLE_MULTIPLIER );
+				(_output_buffer + frame * channels())[chnl]
+					= static_cast<int_sample_t>(AudioEngine::clip(_ab[frame][chnl]) * OUTPUT_SAMPLE_MULTIPLIER);
 			}
 		}
 	}

src/core/audio/AudioFileFlac.cpp

@@ -89,7 +89,7 @@ bool AudioFileFlac::startEncoding()
 	return true;
 }
 
-void AudioFileFlac::writeBuffer(surroundSampleFrame const* _ab, fpp_t const frames, float master_gain)
+void AudioFileFlac::writeBuffer(surroundSampleFrame const* _ab, fpp_t const frames)
 {
 	OutputSettings::BitDepth depth = getOutputSettings().getBitDepth();
 	float clipvalue = std::nextafterf( -1.0f, 0.0f );
@@ -104,15 +104,15 @@ void AudioFileFlac::writeBuffer(surroundSampleFrame const* _ab, fpp_t const fram
 				// Clip the negative side to just above -1.0 in order to prevent it from changing sign
 				// Upstream issue: https://github.com/erikd/libsndfile/issues/309
 				// When this commit is reverted libsndfile-1.0.29 must be made a requirement for FLAC
-				buf[frame*channels() + channel] = std::max(clipvalue, _ab[frame][channel] * master_gain);
+				buf[frame*channels() + channel] = std::max(clipvalue, _ab[frame][channel]);
 			}
 		}
 		sf_writef_float(m_sf, static_cast<float*>(buf.data()), frames);
 	}
 	else // integer PCM encoding
 	{
 		auto buf = std::vector<int_sample_t>(frames * channels());
-		convertToS16(_ab, frames, master_gain, buf.data(), !isLittleEndian());
+		convertToS16(_ab, frames, buf.data(), !isLittleEndian());
 		sf_writef_short(m_sf, static_cast<short*>(buf.data()), frames);
 	}
 

src/core/audio/AudioFileMP3.cpp

@@ -53,21 +53,18 @@ AudioFileMP3::~AudioFileMP3()
 	tearDownEncoder();
 }
 
-void AudioFileMP3::writeBuffer( const surroundSampleFrame * _buf,
-					const fpp_t _frames,
-					const float _master_gain )
+void AudioFileMP3::writeBuffer(const surroundSampleFrame* _buf, const fpp_t _frames)
 {
 	if (_frames < 1)
 	{
 		return;
 	}
 
-	// TODO Why isn't the gain applied by the driver but inside the device?
 	std::vector<float> interleavedDataBuffer(_frames * 2);
 	for (fpp_t i = 0; i < _frames; ++i)
 	{
-		interleavedDataBuffer[2*i] = _buf[i][0] * _master_gain;
-		interleavedDataBuffer[2*i + 1] = _buf[i][1] * _master_gain;
+		interleavedDataBuffer[2*i] = _buf[i][0];
+		interleavedDataBuffer[2*i + 1] = _buf[i][1];
 	}
 
 	size_t minimumBufferSize = 1.25 * _frames + 7200;

src/core/audio/AudioFileOgg.cpp

@@ -185,12 +185,7 @@ bool AudioFileOgg::startEncoding()
 	return true;
 }
 
-
-
-
-void AudioFileOgg::writeBuffer( const surroundSampleFrame * _ab,
-						const fpp_t _frames,
-						const float _master_gain )
+void AudioFileOgg::writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames)
 {
 	int eos = 0;
 
@@ -201,7 +196,7 @@ void AudioFileOgg::writeBuffer( const surroundSampleFrame * _ab,
 	{
 		for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 		{
-			buffer[chnl][frame] = _ab[frame][chnl] * _master_gain;
+			buffer[chnl][frame] = _ab[frame][chnl];
 		}
 	}
 
@@ -258,7 +253,7 @@ void AudioFileOgg::finishEncoding()
 	if( m_ok )
 	{
 		// just for flushing buffers...
-		writeBuffer( nullptr, 0, 0.0f );
+		writeBuffer(nullptr, 0);
 
 		// clean up
 		ogg_stream_clear( &m_os );

src/core/audio/AudioFileWave.cpp

@@ -93,12 +93,7 @@ bool AudioFileWave::startEncoding()
 	return true;
 }
 
-
-
-
-void AudioFileWave::writeBuffer( const surroundSampleFrame * _ab,
-						const fpp_t _frames,
-						const float _master_gain )
+void AudioFileWave::writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames)
 {
 	OutputSettings::BitDepth bitDepth = getOutputSettings().getBitDepth();
 
@@ -109,8 +104,7 @@ void AudioFileWave::writeBuffer( const surroundSampleFrame * _ab,
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
-				buf[frame*channels()+chnl] = _ab[frame][chnl] *
-								_master_gain;
+				buf[frame * channels() + chnl] = _ab[frame][chnl];
 			}
 		}
 		sf_writef_float( m_sf, buf, _frames );
@@ -119,8 +113,7 @@ void AudioFileWave::writeBuffer( const surroundSampleFrame * _ab,
 	else
 	{
 		auto buf = new int_sample_t[_frames * channels()];
-		convertToS16( _ab, _frames, _master_gain, buf,
-							!isLittleEndian() );
+		convertToS16(_ab, _frames, buf, !isLittleEndian());
 
 		sf_writef_short( m_sf, buf, _frames );
 		delete[] buf;

src/core/audio/AudioJack.cpp

@@ -344,13 +344,12 @@ int AudioJack::processCallback(jack_nframes_t nframes)
 	while (done < nframes && !m_stopped)
 	{
 		jack_nframes_t todo = std::min<jack_nframes_t>(nframes - done, m_framesToDoInCurBuf - m_framesDoneInCurBuf);
-		const float gain = audioEngine()->masterGain();
 		for (int c = 0; c < channels(); ++c)
 		{
 			jack_default_audio_sample_t* o = m_tempOutBufs[c];
 			for (jack_nframes_t frame = 0; frame < todo; ++frame)
 			{
-				o[done + frame] = m_outBuf[m_framesDoneInCurBuf + frame][c] * gain;
+				o[done + frame] = m_outBuf[m_framesDoneInCurBuf + frame][c];
 			}
 		}
 		done += todo;

src/core/audio/AudioOss.cpp

@@ -303,7 +303,7 @@ void AudioOss::run()
 			break;
 		}
 
-		int bytes = convertToS16( temp, frames, audioEngine()->masterGain(), outbuf, m_convertEndian );
+		int bytes = convertToS16(temp, frames, outbuf, m_convertEndian);
 		if( write( m_audioFD, outbuf, bytes ) != bytes )
 		{
 			break;

src/core/audio/AudioPortAudio.cpp

@@ -298,15 +298,11 @@ int AudioPortAudio::process_callback(
 		const int min_len = std::min(static_cast<int>(_framesPerBuffer),
 			m_outBufSize - m_outBufPos);
 
-		float master_gain = audioEngine()->masterGain();
-
 		for( fpp_t frame = 0; frame < min_len; ++frame )
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
-				( _outputBuffer + frame * channels() )[chnl] =
-						AudioEngine::clip( m_outBuf[frame][chnl] *
-						master_gain );
+				(_outputBuffer + frame * channels())[chnl] = AudioEngine::clip(m_outBuf[frame][chnl]);
 			}
 		}
 

src/core/audio/AudioPulseAudio.cpp

@@ -278,10 +278,7 @@ void AudioPulseAudio::streamWriteCallback( pa_stream *s, size_t length )
 			m_quit = true;
 			break;
 		}
-		int bytes = convertToS16( temp, frames,
-						audioEngine()->masterGain(),
-						pcmbuf,
-						m_convertEndian );
+		int bytes = convertToS16(temp, frames, pcmbuf, m_convertEndian);
 		if( bytes > 0 )
 		{
 			pa_stream_write( m_s, pcmbuf, bytes, nullptr, 0,

src/core/audio/AudioSampleRecorder.cpp

@@ -85,11 +85,7 @@ std::shared_ptr<const SampleBuffer> AudioSampleRecorder::createSampleBuffer()
 	return std::make_shared<const SampleBuffer>(std::move(bigBuffer), sampleRate());
 }
 
-
-
-
-void AudioSampleRecorder::writeBuffer( const surroundSampleFrame * _ab,
-					const fpp_t _frames, const float )
+void AudioSampleRecorder::writeBuffer(const surroundSampleFrame* _ab, const fpp_t _frames)
 {
 	auto buf = new sampleFrame[_frames];
 	for( fpp_t frame = 0; frame < _frames; ++frame )

src/core/audio/AudioSdl.cpp

@@ -261,13 +261,6 @@ void AudioSdl::sdlAudioCallback( Uint8 * _buf, int _len )
 										  m_currentBufferFramesCount
 										- m_currentBufferFramePos);
 
-		const float gain = audioEngine()->masterGain();
-		for (uint f = 0; f < min_frames_count; f++)
-		{
-			(m_outBuf + m_currentBufferFramePos)[f][0] *= gain;
-			(m_outBuf + m_currentBufferFramePos)[f][1] *= gain;
-		}
-
 		memcpy( _buf, m_outBuf + m_currentBufferFramePos, min_frames_count*sizeof(sampleFrame) );
 		_buf += min_frames_count*sizeof(sampleFrame);
 		_len -= min_frames_count*sizeof(sampleFrame);
@@ -291,10 +284,7 @@ void AudioSdl::sdlAudioCallback( Uint8 * _buf, int _len )
 			m_convertedBufSize = frames * channels()
 						* sizeof( int_sample_t );
 
-			convertToS16( m_outBuf, frames,
-						audioEngine()->masterGain(),
-						(int_sample_t *)m_convertedBuf,
-						m_outConvertEndian );
+			convertToS16(m_outBuf, frames, reinterpret_cast<int_sample_t*>(m_convertedBuf), m_outConvertEndian);
 		}
 		const int min_len = std::min(_len, m_convertedBufSize
 							- m_convertedBufPos);

src/core/audio/AudioSndio.cpp

@@ -167,8 +167,7 @@ void AudioSndio::run()
 			break;
 		}
 
-		uint bytes = convertToS16( temp, frames,
-		    audioEngine()->masterGain(), outbuf, m_convertEndian );
+		uint bytes = convertToS16(temp, frames, outbuf, m_convertEndian);
 		if( sio_write( m_hdl, outbuf, bytes ) != bytes )
 		{
 			break;