Copy audio to UDP/RTP: Opus implementation (2)

sdrbase/audio/audionetsink.cpp

@@ -30,8 +30,11 @@ AudioNetSink::AudioNetSink(QObject *parent) :
     m_codec(CodecL16),
     m_rtpBufferAudio(0),
     m_sampleRate(48000),
+    m_stereo(false),
     m_decimation(1),
     m_decimationCount(0),
+    m_codecInputSize(960),
+    m_codecInputIndex(0),
     m_bufferIndex(0),
     m_port(9998)
 {
@@ -44,8 +47,11 @@ AudioNetSink::AudioNetSink(QObject *parent, int sampleRate, bool stereo) :
     m_codec(CodecL16),
     m_rtpBufferAudio(0),
     m_sampleRate(48000),
+    m_stereo(false),
     m_decimation(1),
     m_decimationCount(0),
+    m_codecInputSize(960),
+    m_codecInputIndex(0),
     m_bufferIndex(0),
     m_port(9998)
 {
@@ -114,9 +120,10 @@ void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
             << " sampleRate: " << sampleRate;
 
     m_codec = codec;
+    m_stereo = stereo;
     m_sampleRate = sampleRate;
 
-    setDecimationFilters();
+    setNewCodecData();
 
     if (m_rtpBufferAudio)
     {
@@ -151,10 +158,28 @@ void AudioNetSink::setDecimation(uint32_t decimation)
 {
     m_decimation = decimation < 1 ? 1 : decimation > 6 ? 6 : decimation;
     qDebug() << "AudioNetSink::setDecimation: " << m_decimation << " from: " << decimation;
-    setDecimationFilters();
+    setNewCodecData();
     m_decimationCount = 0;
 }
 
+void AudioNetSink::setNewCodecData()
+{
+    if (m_codec == CodecOpus)
+    {
+        m_codecInputSize = m_sampleRate / (m_decimation * 50); // 20ms = 1/50s - size is per channel
+        m_codecInputSize = m_codecInputSize > 960 ? 960 : m_codecInputSize; // hard limit of 48 kS/s
+        m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
+        qDebug() << "AudioNetSink::setNewCodecData: CodecOpus:"
+            << " m_codecInputSize: " << m_codecInputSize
+            << " m_codecRatio: " << m_codecRatio
+            << " Fs: " << m_sampleRate/m_decimation
+            << " stereo: " << m_stereo;
+        m_opus.setEncoder(m_sampleRate/m_decimation, m_stereo ? 2 : 1);
+    }
+
+    setDecimationFilters();
+}
+
 void AudioNetSink::setDecimationFilters()
 {
     int decimatedSampleRate = m_sampleRate / m_decimation;
@@ -206,10 +231,6 @@ void AudioNetSink::write(qint16 isample)
                 m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
                 m_bufferIndex = 0;
             }
-        }
-        else if (m_codec == CodecOpus)
-        {
-
         }
         else
         {
@@ -250,7 +271,15 @@ void AudioNetSink::write(qint16 isample)
             break;
         case CodecOpus:
         {
+            if (m_codecInputIndex == m_codecInputSize)
+            {
+                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
+                nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
+                m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
+                m_codecInputIndex = 0;
+            }
 
+            m_opusIn[m_codecInputIndex++] = sample;
         }
             break;
         case CodecL16:
@@ -289,7 +318,7 @@ void AudioNetSink::write(qint16 isample)
                 m_bufferIndex = 0;
             }
 
-            if (m_bufferIndex%2 == 0) {
+            if (m_bufferIndex % 2 == 0) {
                 m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex/2]);
             }
 
@@ -298,6 +327,25 @@ void AudioNetSink::write(qint16 isample)
             m_bufferIndex += 1;
         }
             break;
+        case CodecOpus:
+        {
+            if (m_codecInputIndex == m_codecInputSize)
+            {
+                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
+                if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
+                    qWarning("AudioNetSink::write: CodecOpus mono: unexpected output frame size: %d bytes", nbBytes);
+                }
+                m_bufferIndex = 0;
+                m_codecInputIndex = 0;
+            }
+
+            if (m_codecInputIndex % m_codecRatio == 0) {
+                m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
+            }
+
+            m_opusIn[m_codecInputIndex++] = sample;
+        }
+            break;
         case CodecL16:
         default:
             m_rtpBufferAudio->write((uint8_t *) &sample);
@@ -333,10 +381,43 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
     {
         if (m_bufferIndex >= m_udpBlockSize)
         {
-            m_udpSocket->writeDatagram((const char*)m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
+            m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
             m_bufferIndex = 0;
         }
-        else
+
+        switch(m_codec)
+        {
+        case CodecPCMA:
+        case CodecPCMU:
+        case CodecG722:
+            break; // mono modes - do nothing
+        case CodecOpus:
+        {
+            if (m_codecInputIndex == m_codecInputSize)
+            {
+                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
+                nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
+                m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
+                m_codecInputIndex = 0;
+            }
+
+            m_opusIn[2*m_codecInputIndex]   = lSample;
+            m_opusIn[2*m_codecInputIndex+1] = rSample;
+            m_codecInputIndex++;
+        }
+            break;
+        case CodecL8:
+        {
+            qint8 *p = (qint8*) &m_data[m_bufferIndex];
+            *p = lSample / 256;
+            m_bufferIndex += sizeof(qint8);
+            p = (qint8*) &m_data[m_bufferIndex];
+            *p = rSample / 256;
+            m_bufferIndex += sizeof(qint8);
+        }
+            break;
+        case CodecL16:
+        default:
         {
             qint16 *p = (qint16*) &m_data[m_bufferIndex];
             *p = lSample;
@@ -345,10 +426,50 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
             *p = rSample;
             m_bufferIndex += sizeof(qint16);
         }
+            break;
+        }
     }
     else if (m_type == SinkRTP)
     {
-        m_rtpBufferAudio->write((uint8_t *) &lSample, (uint8_t *) &rSample);
+        switch(m_codec)
+        {
+        case CodecPCMA:
+        case CodecPCMU:
+        case CodecG722:
+            break; // mono modes - do nothing
+        case CodecOpus:
+        {
+            if (m_codecInputIndex == m_codecInputSize)
+            {
+                int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
+                if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
+                    qWarning("AudioNetSink::write: CodecOpus stereo: unexpected output frame size: %d bytes", nbBytes);
+                }
+                m_bufferIndex = 0;
+                m_codecInputIndex = 0;
+            }
+
+            if (m_codecInputIndex % m_codecRatio == 0) {
+                m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
+            }
+
+            m_opusIn[2*m_codecInputIndex]   = lSample;
+            m_opusIn[2*m_codecInputIndex+1] = rSample;
+            m_codecInputIndex++;
+        }
+            break;
+        case CodecL8:
+        {
+            qint8 pl = lSample / 256;
+            qint8 pr = rSample / 256;
+            m_rtpBufferAudio->write((uint8_t *) &pl, (uint8_t *) &pr);
+        }
+            break;
+        case CodecL16:
+        default:
+            m_rtpBufferAudio->write((uint8_t *) &lSample, (uint8_t *) &rSample);
+            break;
+        }
     }
 }
 

sdrbase/audio/audionetsink.h

@@ -22,6 +22,7 @@
 #include "audiofilter.h"
 #include "audiocompressor.h"
 #include "audiog722.h"
+#include "audioopus.h"
 #include "export.h"
 
 #include <QObject>
@@ -72,9 +73,11 @@ public:
     static const int m_dataBlockSize = 65536; // room for G722 conversion (64000 = 12800*5 largest to date)
     static const int m_g722BlockSize = 12800; // number of resulting G722 bytes (80*20ms frames)
     static const int m_opusBlockSize = 960*2; // provision for 20ms of 2 int16 channels at 48 kS/s
+    static const int m_opusOutputSize = 160;  // output frame: 20ms of 8 bit data @ 64 kbits/s = 160 bytes
 
 protected:
-    void setDecimationFilters();
+    void setNewCodecData();       // actions to take when changes affecting codec dependent data occurs
+    void setDecimationFilters();  // set decimation filters limits depending on effective sample rate and codec
 
     SinkType m_type;
     Codec m_codec;
@@ -82,18 +85,20 @@ protected:
     RTPSink *m_rtpBufferAudio;
     AudioCompressor m_audioCompressor;
     AudioG722 m_g722;
+    AudioOpus m_opus;
     AudioFilter m_audioFilter;
     int m_sampleRate;
+    bool m_stereo;
     uint32_t m_decimation;
     uint32_t m_decimationCount;
     char m_data[m_dataBlockSize];
-    uint16_t m_opusIn[m_opusBlockSize];
+    int16_t m_opusIn[m_opusBlockSize];
+    int m_codecInputSize;  // codec input block size - for codecs with actual encoding (Opus only for now)
+    int m_codecInputIndex; // codec input block fill index
+    int m_codecRatio;      // codec compression ratio
     unsigned int m_bufferIndex;
     QHostAddress m_address;
     unsigned int m_port;
 };
 
-
-
-
 #endif /* SDRBASE_AUDIO_AUDIONETSINK_H_ */

sdrbase/audio/audioopus.cpp

@@ -2,6 +2,9 @@
 // Copyright (C) 2019 F4EXB                                                      //
 // written by Edouard Griffiths                                                  //
 //                                                                               //
+// In this version we will use a fixed constant bit rate of 64kbit/s.            //
+// With a frame time of 20ms the encoder output size is always 160 bytes.        //
+//                                                                               //
 // 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 as version 3 of the License, or                  //
@@ -69,6 +72,15 @@ void AudioOpus::setEncoder(int32_t fs, int nChannels)
         m_encoderOK = false;
         return;
     }
+
+    error = opus_encoder_ctl(m_encoderState, OPUS_SET_VBR(0)); // force constant bit rate
+
+    if (error != OPUS_OK)
+    {
+        qWarning("AudioOpus::setEncoder: set constant bitrate error: %s", opus_strerror(error));
+        m_encoderOK = false;
+        return;
+    }
 }
 
 int AudioOpus::encode(int frameSize, int16_t *in, uint8_t *out)
@@ -77,7 +89,7 @@ int AudioOpus::encode(int frameSize, int16_t *in, uint8_t *out)
 
     if (nbBytes < 0)
     {
-        qWarning("AudioOpus::encode failed: %s\n", opus_strerror(nbBytes));
+        qWarning("AudioOpus::encode failed: %s", opus_strerror(nbBytes));
         return 0;
     }
     else

sdrbase/util/rtpsink.cpp

@@ -113,8 +113,8 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
     case PayloadOpus:
         m_sampleBytes = 1;
         m_rtpSession.SetDefaultPayloadType(101);
-        m_packetSamples = 160; // Fixed 20ms @ 64 kbits/s packet samples
+        m_packetSamples = 960; // Fixed 20ms @ 48 kHz as per https://tools.ietf.org/html/rfc7587
-        timestampinc = 160;    // 1 channel
+        timestampinc = 960;    // and per single channel
         break;
     case PayloadL16Mono:
     default:

sdrgui/audio.md

@@ -67,7 +67,7 @@ This is the codec applied before sending the stream via UDP. The following are a
   - `PCMA`: A-law 8 bit PCM (requires 8000 Hz sample rate mono)
   - `PCMU`: Mu-law 8 bit PCM (requires 8000 Hz sample rate mono)
   - `G722`: G722 64 kbit/s (requires 16000 Hz sample rate mono)
-  - `OPUS` : Opus 64 kbit/s
+  - `OPUS` : Opus constant 64 kbit/s (requires 48, 24, 16 or 12 kHz sample rates)
 
 <h3>1.10 SDP string</h3>
 

sdrgui/gui/audiodialog.cpp

@@ -350,4 +350,11 @@ void AudioDialogX::check()
             QMessageBox::information(this, tr("Message"), tr("G722 must be 16000 Hz single channel"));
         }
     }
+    else if (m_outputDeviceInfo.udpChannelCodec == AudioOutput::UDPCodecOpus)
+    {
+        int effectiveSampleRate = m_outputDeviceInfo.sampleRate/decimationFactor;
+        if ((effectiveSampleRate != 48000) || (effectiveSampleRate != 24000) || (effectiveSampleRate != 16000) || (effectiveSampleRate != 12000)) {
+            QMessageBox::information(this, tr("Message"), tr("Opus takes only 48, 24, 16 or 12 kHz sample rates"));
+        }
+    }
 }