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Audio UDP/RTP: implemented audio decimation in the net sink

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This commit is contained in:
f4exb 2019-02-14 17:21:14 +01:00
parent 8582d52739
commit 475d8647f0
5 changed files with 66 additions and 33 deletions
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1
sdrbase/audio/audiodevicemanager.cpp
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@ -614,6 +614,7 @@ void AudioDeviceManager::setOutputDeviceInfo(int outputDeviceIndex, const Output
audioOutput->setUdpUseRTP(deviceInfo.udpUseRTP);
audioOutput->setUdpChannelMode(deviceInfo.udpChannelMode);
audioOutput->setUdpChannelFormat(deviceInfo.udpChannelCodec, deviceInfo.udpChannelMode == AudioOutput::UDPChannelStereo, deviceInfo.sampleRate);
audioOutput->setUdpDecimation(deviceInfo.decimationFactor);
qDebug("AudioDeviceManager::setOutputDeviceInfo: index: %d device: %s updated",
outputDeviceIndex, qPrintable(deviceName));

84
sdrbase/audio/audionetsink.cpp
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@ -27,6 +27,8 @@ AudioNetSink::AudioNetSink(QObject *parent) :
m_type(SinkUDP),
m_codec(CodecL16),
m_rtpBufferAudio(0),
m_decimation(1),
m_decimationCount(0),
m_bufferIndex(0),
m_port(9998)
{
@ -38,6 +40,8 @@ AudioNetSink::AudioNetSink(QObject *parent, int sampleRate, bool stereo) :
m_type(SinkUDP),
m_codec(CodecL16),
m_rtpBufferAudio(0),
m_decimation(1),
m_decimationCount(0),
m_bufferIndex(0),
m_port(9998)
{
@ -130,8 +134,33 @@ void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
}
}
void AudioNetSink::write(qint16 sample)
void AudioNetSink::setDecimation(uint32_t decimation)
{
m_decimation = decimation < 1 ? 1 : decimation > 6 ? 6 : decimation;
qDebug() << "AudioNetSink::setDecimation: " << m_decimation << " from: " << decimation;
m_decimationCount = 0;
}
void AudioNetSink::write(qint16 isample)
{
qint16& sample = isample;
if (m_decimation > 1)
{
float lpSample = m_audioFilter.runLP(sample / 32768.0f);
if (m_decimationCount >= m_decimation - 1)
{
sample = lpSample * 32768.0f;
m_decimationCount = 0;
}
else
{
m_decimationCount++;
return;
}
}
if (m_type == SinkUDP)
{
if (m_bufferIndex >= m_udpBlockSize)
@ -194,8 +223,29 @@ void AudioNetSink::write(qint16 sample)
}
}
void AudioNetSink::write(qint16 lSample, qint16 rSample)
void AudioNetSink::write(qint16 ilSample, qint16 irSample)
{
qint16& lSample = ilSample;
qint16& rSample = irSample;
if (m_decimation > 1)
{
float lpLSample = m_audioFilter.runLP(lSample / 32768.0f);
float lpRSample = m_audioFilter.runLP(rSample / 32768.0f);
if (m_decimationCount >= m_decimation - 1)
{
lSample = lpLSample * 32768.0f;
rSample = lpRSample * 32768.0f;
m_decimationCount = 0;
}
else
{
m_decimationCount++;
return;
}
}
if (m_type == SinkUDP)
{
if (m_bufferIndex >= m_udpBlockSize)
@ -219,36 +269,6 @@ void AudioNetSink::write(qint16 lSample, qint16 rSample)
}
}
//void AudioNetSink::write(AudioSample* samples, uint32_t numSamples)
//{
// if (m_type == SinkUDP)
// {
// int samplesIndex = 0;
//
// if (m_bufferIndex + numSamples*sizeof(AudioSample) >= m_udpBlockSize) // fill remainder of buffer and send it
// {
// memcpy(&m_data[m_bufferIndex], &samples[samplesIndex], m_udpBlockSize - m_bufferIndex); // fill remainder of buffer
// m_udpSocket->writeDatagram((const char*)m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
// m_bufferIndex = 0;
// samplesIndex += (m_udpBlockSize - m_bufferIndex) / sizeof(AudioSample);
// numSamples -= (m_udpBlockSize - m_bufferIndex) / sizeof(AudioSample);
// }
//
// while (numSamples > m_udpBlockSize/sizeof(AudioSample)) // send directly from input without buffering
// {
// m_udpSocket->writeDatagram((const char*)&samples[samplesIndex], (qint64 ) m_udpBlockSize, m_address, m_port);
// samplesIndex += m_udpBlockSize/sizeof(AudioSample);
// numSamples -= m_udpBlockSize/sizeof(AudioSample);
// }
//
// memcpy(&m_data[m_bufferIndex], &samples[samplesIndex], numSamples*sizeof(AudioSample));
// }
// else if (m_type == SinkRTP)
// {
// m_rtpBufferAudio->write((uint8_t *) samples, numSamples*2); // 2 x 16 bit sample
// }
//}
void AudioNetSink::moveToThread(QThread *thread)
{
m_udpSocket->moveToThread(thread);

6
sdrbase/audio/audionetsink.h
View File

@ -19,6 +19,7 @@
#define SDRBASE_AUDIO_AUDIONETSINK_H_
#include "dsp/dsptypes.h"
#include "audiofilter.h"
#include "audiocompressor.h"
#include "export.h"
@ -54,10 +55,10 @@ public:
void addDestination(const QString& address, uint16_t port);
void deleteDestination(const QString& address, uint16_t port);
void setParameters(Codec codec, bool stereo, int sampleRate);
void setDecimation(uint32_t decimation);
void write(qint16 sample);
void write(qint16 lSample, qint16 rSample);
//void write(AudioSample* samples, uint32_t numSamples);
bool isRTPCapable() const;
bool selectType(SinkType type);
@ -72,6 +73,9 @@ protected:
QUdpSocket *m_udpSocket;
RTPSink *m_rtpBufferAudio;
AudioCompressor m_audioCompressor;
AudioFilter m_audioFilter;
uint32_t m_decimation;
uint32_t m_decimationCount;
char m_data[65536];
unsigned int m_bufferIndex;
QHostAddress m_address;

7
sdrbase/audio/audiooutput.cpp
View File

@ -210,6 +210,13 @@ void AudioOutput::setUdpChannelFormat(UDPChannelCodec udpChannelCodec, bool ster
}
}
void AudioOutput::setUdpDecimation(uint32_t decimation)
{
if (m_audioNetSink) {
m_audioNetSink->setDecimation(decimation);
}
}
qint64 AudioOutput::readData(char* data, qint64 maxLen)
{
//qDebug("AudioOutput::readData: %lld", maxLen);

1
sdrbase/audio/audiooutput.h
View File

@ -67,6 +67,7 @@ public:
void setUdpUseRTP(bool useRTP);
void setUdpChannelMode(UDPChannelMode udpChannelMode);
void setUdpChannelFormat(UDPChannelCodec udpChannelCodec, bool stereo, int sampleRate);
void setUdpDecimation(uint32_t decimation);
private:
QMutex m_mutex;