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https://github.com/f4exb/sdrangel.git
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DSD demod: DMR integration debug (2). Added eye diagram capability by delaying Q signal by 20 samples (2400 baud)
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f4exb
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f167241da7
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90a5b75e91
@ -123,10 +123,11 @@ void DSDDMRVoice::process()
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case 9:
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processSlot9(symbolIndex);
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break;
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case 10:
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case 10: // this is the post-process case
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postProcess(symbolIndex);
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break;
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default:
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m_dsdDecoder->m_fsmState = DSDDecoder::DSDLookForSync;
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break;
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}
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@ -172,10 +173,12 @@ void DSDDMRVoice::preProcess()
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void DSDDMRVoice::postProcess(int symbolIndex)
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{
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//fprintf(stderr, "DSDDMRVoice::postProcess: m_symbolIndex: %d", m_symbolIndex);
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m_dsdDecoder->getDibit(); // get dibit from symbol but do nothing with it
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if (symbolIndex == 54+12+54-1) // very last symbol -> go back to search sync state
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{
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fprintf(stderr, "\nDSDDMRVoice::postProcess: end of frame\n");
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m_dsdDecoder->m_fsmState = DSDDecoder::DSDLookForSync;
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}
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}
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@ -59,6 +59,12 @@ public:
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private:
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int getSlotIndex(int symbolInMajorBlockIndex) //!< calculates current slot index and returns symbol index in the slot
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{
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if (m_majorBlock > 5) // this is the post-process case
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{
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m_slotIndex = 10;
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return symbolInMajorBlockIndex;
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}
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if (symbolInMajorBlockIndex < 54)
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{
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m_slotIndex = 0;
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@ -109,11 +115,6 @@ private:
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m_slotIndex = 9;
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return symbolInMajorBlockIndex - 54+12+36+18+24+18+36+12+54;
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}
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else if (symbolInMajorBlockIndex < 54+12+36+18+24+18+36+12+54+24+54+12+54)
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{
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m_slotIndex = 10; // dummy slot for last skipped symbols
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return symbolInMajorBlockIndex - 54+12+36+18+24+18+36+12+54+24;
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}
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else // cannot go there if using this function in its valid context (input is a remainder of division by 288)
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{
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m_slotIndex = -1; // invalid slot
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@ -59,6 +59,9 @@ DSDDemod::DSDDemod(SampleSink* sampleSink) :
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_sampleBuffer = new qint16[1<<17]; // 128 kS
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m_sampleBufferIndex = 0;
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m_movingAverage.resize(16, 0);
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DSPEngine::instance()->addAudioSink(&m_audioFifo);
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@ -66,6 +69,7 @@ DSDDemod::DSDDemod(SampleSink* sampleSink) :
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DSDDemod::~DSDDemod()
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{
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delete[] m_sampleBuffer;
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DSPEngine::instance()->removeAudioSink(&m_audioFifo);
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}
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@ -102,7 +106,7 @@ void DSDDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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qint16 sample;
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qint16 sample, delayedSample;
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m_magsq = ((ci.real()*ci.real() + ci.imag()*ci.imag())) / (Real) (1<<30);
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m_movingAverage.feed(m_magsq);
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@ -135,40 +139,53 @@ void DSDDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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sample = 0;
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}
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Sample s(sample, 0.0);
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if (m_sampleBufferIndex < (1<<17)) {
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m_sampleBufferIndex++;
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} else {
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m_sampleBufferIndex = 0;
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}
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m_sampleBuffer[m_sampleBufferIndex] = sample;
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if (m_sampleBufferIndex < 20) {
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delayedSample = m_sampleBuffer[(1<<17) - 20 + m_sampleBufferIndex];
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} else {
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delayedSample = m_sampleBuffer[m_sampleBufferIndex - 20];
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}
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Sample s(sample, delayedSample); // I=signal, Q=signal delayed by 20 samples (2400 baud: lowest rate)
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m_scopeSampleBuffer.push_back(s);
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m_dsdDecoder.pushSample(sample);
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// if (m_running.m_audioMute)
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// {
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// m_audioBuffer[m_audioBufferFill].l = 0;
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// m_audioBuffer[m_audioBufferFill].r = 0;
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// }
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// else
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// {
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// m_audioBuffer[m_audioBufferFill].l = sample;
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// m_audioBuffer[m_audioBufferFill].r = sample;
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// }
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// if (m_running.m_audioMute)
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// {
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// m_audioBuffer[m_audioBufferFill].l = 0;
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// m_audioBuffer[m_audioBufferFill].r = 0;
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// }
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// else
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// {
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// m_audioBuffer[m_audioBufferFill].l = (sample * m_running.m_volume) / 100;
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// m_audioBuffer[m_audioBufferFill].r = (sample * m_running.m_volume) / 100;
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// }
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//
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// ++m_audioBufferFill;
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// ++m_audioBufferFill;
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//
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// if (m_audioBufferFill >= m_audioBuffer.size())
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// {
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// uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
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// if (m_audioBufferFill >= m_audioBuffer.size())
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// {
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// uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
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//
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// if (res != m_audioBufferFill)
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// {
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// qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
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// }
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// if (res != m_audioBufferFill)
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// {
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// qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
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// }
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//
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// m_audioBufferFill = 0;
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// }
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// m_audioBufferFill = 0;
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// }
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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// if (m_audioBufferFill > 0)
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// {
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// uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
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@ -182,18 +199,23 @@ void DSDDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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// }
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int nbAudioSamples;
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short *audio = m_dsdDecoder.getAudio(nbAudioSamples);
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short *dsdAudio = m_dsdDecoder.getAudio(nbAudioSamples);
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if (nbAudioSamples > 0)
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], nbAudioSamples, 10);
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if (res != nbAudioSamples)
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{
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qDebug("NFMDemod::feed: %u/%u tail samples written", res, nbAudioSamples);
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}
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uint res = m_audioFifo.write((const quint8*) dsdAudio, nbAudioSamples, 10);
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qDebug("DSDDemod::feed: written %d audio samples (%d)", res, nbAudioSamples);
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m_dsdDecoder.resetAudio();
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// qDebug("\nDSDDemod::feed: got %d audio samples (%lu)", nbAudioSamples, m_audioBuffer.size());
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}
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// if (nbAudioSamples >= m_audioBuffer.size())
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// {
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// uint res = m_audioFifo.write((const quint8*) dsdAudio, nbAudioSamples, 10);
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// qDebug("DSDDemod::feed: written %d audio samples (%d)", res, nbAudioSamples);
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// m_dsdDecoder.resetAudio();
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// }
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if ((m_scope != 0) && (m_scopeEnabled))
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{
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m_scope->feed(m_scopeSampleBuffer.begin(), m_scopeSampleBuffer.end(), true); // true = real samples for what it's worth
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@ -172,6 +172,8 @@ private:
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SampleVector m_scopeSampleBuffer;
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AudioVector m_audioBuffer;
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uint m_audioBufferFill;
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qint16 *m_sampleBuffer; //!< samples ring buffer
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int m_sampleBufferIndex;
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AudioFifo m_audioFifo;
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SampleSink* m_scope;
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@ -151,7 +151,7 @@ void FileSourceGui::handleSourceMessages()
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while ((message = m_sampleSource->getOutputMessageQueueToGUI()->pop()) != 0)
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{
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qDebug("FileSourceGui::handleSourceMessages: message: %s", message->getIdentifier());
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//qDebug("FileSourceGui::handleSourceMessages: message: %s", message->getIdentifier());
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if (handleMessage(*message))
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{
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