MBE audio interpolator and filter: use a volume setting for fixed values

f4exb · Jun 24, 2018 · 8a1d78b · 8a1d78b
1 parent 4b2b594
commit 8a1d78b
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Showing 3 changed files with 23 additions and 16 deletions.
diff --git a/dsd_decoder.cpp b/dsd_decoder.cpp
@@ -249,17 +249,21 @@ void DSDDecoder::setAudioGain(float gain)
     else if (m_opts.audio_gain == 0.0f)
     {
         m_dsdLogger.log("Enabling audio out auto-gain\n");
-    	m_mbeDecoder1.setAudioGain(25);
-    	m_mbeDecoder1.setAutoGain(true);
+        m_mbeDecoder1.setAudioGain(25);
+        m_mbeDecoder1.setVolume(1.0f);
+        m_mbeDecoder1.setAutoGain(true);
         m_mbeDecoder2.setAudioGain(25);
+        m_mbeDecoder2.setVolume(1.0f);
         m_mbeDecoder2.setAutoGain(true);
     }
     else
     {
         m_dsdLogger.log("Setting audio out gain to %f\n", m_opts.audio_gain);
         m_mbeDecoder1.setAudioGain(m_opts.audio_gain);
+        m_mbeDecoder1.setVolume(m_opts.audio_gain);
         m_mbeDecoder1.setAutoGain(false);
         m_mbeDecoder2.setAudioGain(m_opts.audio_gain);
+        m_mbeDecoder2.setVolume(m_opts.audio_gain);
         m_mbeDecoder2.setAutoGain(false);
     }
 }
@@ -1396,7 +1400,7 @@ void DSDDecoder::formatStatusText(char *statusText)
             // 1    2    2    3    3    4    4    5    5    6    6    7    7    8
             // 5....0....5....0....5....0....5....0....5....0....5....0....5....0..
             // NXD>RC cc mm llllll ssss
-            sprintf(&statusText[15], "NXD>RC %02d %02X %06X %02X", 
+            sprintf(&statusText[15], "NXD>RC %02d %02X %06X %02X",
                 getNXDNDecoder().getRAN(),
                 getNXDNDecoder().getMessageType(),
                 getNXDNDecoder().getLocationId(),

diff --git a/dsd_mbe.cpp b/dsd_mbe.cpp
@@ -49,6 +49,7 @@ DSDMBEDecoder::DSDMBEDecoder(DSDDecoder *dsdDecoder) :
     m_audio_out_idx2 = 0;
 
     m_aout_gain = 25;
+    m_volume = 1.0f;
     m_auto_gain = true;
     m_stereo = false;
     m_channels = 3; // both channels by default if stereo is set
@@ -238,24 +239,24 @@ void DSDMBEDecoder::processAudio()
         }
 
         gaindelta /= (float) 160;
+
+        // adjust output gain
+        m_audio_out_temp_buf_p = m_audio_out_temp_buf;
+
+        for (n = 0; n < 160; n++)
+        {
+            *m_audio_out_temp_buf_p = (m_aout_gain
+                    + ((float) n * gaindelta)) * (*m_audio_out_temp_buf_p);
+            m_audio_out_temp_buf_p++;
+        }
+
+        m_aout_gain += ((float) 160 * gaindelta);
     }
     else
     {
         gaindelta = (float) 0;
     }
 
-	// adjust output gain
-	m_audio_out_temp_buf_p = m_audio_out_temp_buf;
-
-	for (n = 0; n < 160; n++)
-	{
-		*m_audio_out_temp_buf_p = (m_aout_gain
-				+ ((float) n * gaindelta)) * (*m_audio_out_temp_buf_p);
-		m_audio_out_temp_buf_p++;
-	}
-
-	m_aout_gain += ((float) 160 * gaindelta);
-
     // copy audio data to output buffer and upsample if necessary
     m_audio_out_temp_buf_p = m_audio_out_temp_buf;
 
@@ -369,7 +370,7 @@ void DSDMBEDecoder::upsample(int upsampling, float invalue)
 //    outbuf1--;
 //    c = *outbuf1;
     c = m_upsamplerLastValue;
-    d = m_upsamplingFilter.usesHP() ? m_upsamplingFilter.runHP(invalue) : invalue;
+    d = (m_upsamplingFilter.usesHP() ? m_upsamplingFilter.runHP(invalue) : invalue)*m_volume;
     // basic triangle interpolation
 //    outbuf1++;
     if (upsampling == 2)

diff --git a/dsd_mbe.h b/dsd_mbe.h
@@ -49,6 +49,7 @@ class DSDMBEDecoder
 
     void setAudioGain(float aout_gain) { m_aout_gain = aout_gain; }
     void setAutoGain(bool auto_gain) { m_auto_gain = auto_gain; }
+    void setVolume(float volume) { m_volume = volume; }
     void setStereo(bool stereo) { m_stereo = stereo; }
     void setChannels(unsigned char channels) { m_channels = channels % 4; }
     void setUpsamplingFactor(int upsample) { m_upsample = upsample; }
@@ -87,6 +88,7 @@ class DSDMBEDecoder
     int   m_audio_out_idx2;
 
     float m_aout_gain;
+    float m_volume;
     bool m_auto_gain;
     int m_upsample;            //!< upsampling factor
     bool m_stereo;             //!< double each audio sample to produce L+R channels