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sdrangel/plugins/channelrx/demoddab/dabdemodsink.cpp

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019 Edouard Griffiths, F4EXB //
// Copyright (C) 2021 Jon Beniston, M7RCE //
// //
// 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 //
// (at your option) any later version. //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <QDebug>
#include <complex.h>
#include "dsp/dspengine.h"
#include "dsp/datafifo.h"
#include "util/db.h"
#include "pipes/pipeendpoint.h"
#include "maincore.h"
#include "dabdemod.h"
#include "dabdemodsink.h"
// Callbacks from DAB library
void syncHandler(bool value, void *ctx)
{
(void)value;
(void)ctx;
}
void systemDataHandler(bool sync, int16_t snr, int32_t freqOffset, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->systemData(sync, snr, freqOffset);
}
void ensembleNameHandler(const char *name, int32_t id, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->ensembleName(QString::fromUtf8(name), id);
}
void programNameHandler(const char *name, int32_t id, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->programName(QString::fromUtf8(name), id);
}
void fibQualityHandler(int16_t percent, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->fibQuality(percent);
}
void audioHandler(int16_t *buffer, int size, int samplerate, bool stereo, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->audio(buffer, size, samplerate, stereo);
}
void dataHandler(const char *data, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->data(QString::fromUtf8(data));
}
void byteHandler(uint8_t *data, int16_t a, uint8_t b, void *ctx)
{
(void)data;
(void)a;
(void)b;
(void)ctx;
}
// Note: North America has different table
static const char *dabProgramType[] =
{
"No programme type",
"News",
"Current Affairs",
"Information",
"Sport",
"Education",
"Drama",
"Culture",
"Science",
"Varied",
"Pop Music",
"Rock Music",
"Easy Listening Music",
"Light Classical",
"Serious Classical",
"Other Music",
"Weather/meteorology",
"Finance/Business",
"Children's programmes",
"Social Affairs",
"Religion",
"Phone In",
"Travel",
"Leisure",
"Jazz Music",
"Country Music",
"National Music",
"Oldies Music",
"Folk Music",
"Documentary",
"Not used",
"Not used",
};
static const char *dabLanguageCode[] =
{
"Unknown",
"Albanian",
"Breton",
"Catalan",
"Croatian",
"Welsh",
"Czech",
"Danish",
"German",
"English",
"Spanish",
"Esperanto",
"Estonian",
"Basque",
"Faroese",
"French",
"Frisian",
"Irish",
"Gaelic",
"Galician",
"Icelandic",
"Italian",
"Sami",
"Latin",
"Latvian",
"Luxembourgian",
"Lithuanian",
"Hungarian",
"Maltese",
"Dutch",
"Norwegian",
"Occitan",
"Polish",
"Portuguese",
"Romanian",
"Romansh",
"Serbian",
"Slovak",
"Slovene",
"Finnish",
"Swedish",
"Turkish",
"Flemish",
"Walloon",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Background sound",
"Reserved",
"Reserved",
"Reserved",
"Reserved",
"Zulu",
"Vietnamese",
"Uzbek",
"Urdu",
"Ukranian",
"Thai",
"Telugu",
"Tatar",
"Tamil",
"Tadzhik",
"Swahili",
"Sranan Tongo",
"Somali",
"Sinhalese",
"Shona",
"Serbo-Croat",
"Rusyn",
"Russian",
"Quechua",
"Pushtu",
"Punjabi",
"Persian",
"Papiamento",
"Oriya",
"Nepali",
"Ndebele",
"Marathi",
"Moldavian",
"Malaysian",
"Malagasay",
"Macedonian",
"Laotian",
"Korean",
"Khmer",
"Kazakh",
"Kannada",
"Japanese",
"Indonesian",
"Hindi",
"Hebrew",
"Hausa",
"Gurani",
"Gujurati",
"Greek",
"Georgian",
"Fulani",
"Dari",
"Chuvash",
"Chinese",
"Burmese",
"Bulgarian",
"Bengali",
"Belorussian",
"Bambora",
"Azerbaijani",
"Assamese",
"Armenian",
"Arabic",
"Amharic",
};
void programDataHandler(audiodata *data, void *ctx)
{
QString audio;
if (data->ASCTy == 0)
audio = "DAB";
else if (data->ASCTy == 63)
audio = "DAB+";
else
audio = "Unknown";
QString language = "";
if ((data->language < 0x80) && (data->language >= 0))
language = dabLanguageCode[data->language & 0x7f];
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->programData(data->bitRate, audio, language, dabProgramType[data->programType & 0x1f]);
}
void programQualityHandler(int16_t frames, int16_t rs, int16_t aac, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->programQuality(frames, rs, aac);
}
void motDataHandler(uint8_t *data, int len, const char *filename, int contentsubType, void *ctx)
{
DABDemodSink *sink = (DABDemodSink *)ctx;
sink->motData(data, len, QString::fromUtf8(filename), contentsubType);
}
void DABDemodSink::systemData(bool sync, int16_t snr, int32_t freqOffset)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABSystemData *msg = DABDemod::MsgDABSystemData::create(sync, snr, freqOffset);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::ensembleName(const QString& name, int id)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABEnsembleName *msg = DABDemod::MsgDABEnsembleName::create(name, id);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::programName(const QString& name, int id)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABProgramName *msg = DABDemod::MsgDABProgramName::create(name, id);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::programData(int bitrate, const QString& audio, const QString& language, const QString& programType)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABProgramData *msg = DABDemod::MsgDABProgramData::create(bitrate, audio, language, programType);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::fibQuality(int16_t percent)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABFIBQuality *msg = DABDemod::MsgDABFIBQuality::create(percent);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::programQuality(int16_t frames, int16_t rs, int16_t aac)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABProgramQuality *msg = DABDemod::MsgDABProgramQuality::create(frames, rs, aac);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::data(const QString& data)
{
if (getMessageQueueToChannel())
{
DABDemod::MsgDABData *msg = DABDemod::MsgDABData::create(data);
getMessageQueueToChannel()->push(msg);
}
}
void DABDemodSink::motData(const uint8_t *data, int len, const QString& filename, int contentSubType)
{
if (getMessageQueueToChannel())
{
QByteArray byteArray((const char *)data, len);
DABDemod::MsgDABMOTData *msg = DABDemod::MsgDABMOTData::create(byteArray, filename, contentSubType);
getMessageQueueToChannel()->push(msg);
}
}
static int16_t scale(int16_t sample, float factor)
{
int32_t prod = (int32_t)(((int32_t)sample) * factor);
prod = std::min(prod, 32767);
prod = std::max(prod, -32768);
return (int16_t)prod;
}
void DABDemodSink::audio(int16_t *buffer, int size, int samplerate, bool stereo)
{
(void)stereo;
(void)samplerate;
if (samplerate != m_dabAudioSampleRate)
{
applyDABAudioSampleRate(samplerate);
if (getMessageQueueToChannel())
{
DABDemod::MsgDABSampleRate *msg = DABDemod::MsgDABSampleRate::create(samplerate);
getMessageQueueToChannel()->push(msg);
}
}
// buffer is always 2 channels
for (int i = 0; i < size; i+=2)
{
Complex ci, ca;
if (!m_settings.m_audioMute)
{
ci.real(buffer[i]);
ci.imag(buffer[i+1]);
}
else
{
ci.real(0.0f);
ci.imag(0.0f);
}
if (m_audioInterpolatorDistance < 1.0f) // interpolate
{
while (!m_audioInterpolator.interpolate(&m_audioInterpolatorDistanceRemain, ci, &ca))
{
processOneAudioSample(ca);
m_audioInterpolatorDistanceRemain += m_audioInterpolatorDistance;
}
}
else // decimate
{
if (m_audioInterpolator.decimate(&m_audioInterpolatorDistanceRemain, ci, &ca))
{
processOneAudioSample(ca);
m_audioInterpolatorDistanceRemain += m_audioInterpolatorDistance;
}
}
}
}
void DABDemodSink::reset()
{
dabReset(m_dab);
}
void DABDemodSink::resetService()
{
dabReset_msc(m_dab);
}
void DABDemodSink::processOneAudioSample(Complex &ci)
{
float factor = m_settings.m_volume / 5.0f; // Should this be 5 or 10? 5 allows some positive gain
qint16 l = scale(ci.real(), factor);
qint16 r = scale(ci.real(), factor);
m_audioBuffer[m_audioBufferFill].l = l;
m_audioBuffer[m_audioBufferFill].r = r;
++m_audioBufferFill;
if (m_audioBufferFill >= m_audioBuffer.size())
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
if (res != m_audioBufferFill)
{
qDebug("DABDemodSink::audio: %u/%u audio samples written", res, m_audioBufferFill);
m_audioFifo.clear();
}
m_audioBufferFill = 0;
}
m_demodBuffer[m_demodBufferFill++] = l; // FIXME: What about right channel?
if (m_demodBufferFill >= m_demodBuffer.size())
{
QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
if (dataFifos)
{
QList<DataFifo*>::iterator it = dataFifos->begin();
for (; it != dataFifos->end(); ++it) {
(*it)->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16));
}
}
m_demodBufferFill = 0;
}
}
DABDemodSink::DABDemodSink(DABDemod *packetDemod) :
m_dabDemod(packetDemod),
m_audioSampleRate(48000),
m_dabAudioSampleRate(10000), // Unused value to begin with
m_channelSampleRate(DABDEMOD_CHANNEL_SAMPLE_RATE),
m_channelFrequencyOffset(0),
m_magsqSum(0.0f),
m_magsqPeak(0.0f),
m_magsqCount(0),
m_messageQueueToChannel(nullptr),
m_audioFifo(48000)
{
m_audioBuffer.resize(1<<14);
m_audioBufferFill = 0;
m_magsq = 0.0;
m_demodBuffer.resize(1<<12);
m_demodBufferFill = 0;
applySettings(m_settings, true);
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
m_api.dabMode = 1; // Latest DAB spec only has mode 1
m_api.syncsignal_Handler = syncHandler;
m_api.systemdata_Handler = systemDataHandler;
m_api.ensemblename_Handler = ensembleNameHandler;
m_api.programname_Handler = programNameHandler;
m_api.fib_quality_Handler = fibQualityHandler;
m_api.audioOut_Handler = audioHandler;
m_api.dataOut_Handler = dataHandler;
m_api.bytesOut_Handler = byteHandler;
m_api.programdata_Handler = programDataHandler;
m_api.program_quality_Handler = programQualityHandler;
m_api.motdata_Handler = motDataHandler;
m_api.tii_data_Handler = nullptr;
m_api.timeHandler = nullptr;
m_dab = dabInit(&m_device,
&m_api,
nullptr,
nullptr,
this);
dabStartProcessing(m_dab);
}
DABDemodSink::~DABDemodSink()
{
dabExit(m_dab);
}
void DABDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
{
Complex ci;
for (SampleVector::const_iterator it = begin; it != end; ++it)
{
Complex c(it->real(), it->imag());
c *= m_nco.nextIQ();
if (m_interpolatorDistance == 1.0f)
{
processOneSample(c);
}
else if (m_interpolatorDistance < 1.0f) // interpolate
{
while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
else // decimate
{
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
}
}
void DABDemodSink::processOneSample(Complex &ci)
{
// Calculate average and peak levels for level meter
double magsqRaw = ci.real()*ci.real() + ci.imag()*ci.imag();
Real magsq = (Real)(magsqRaw / (SDR_RX_SCALED*SDR_RX_SCALED));
m_movingAverage(magsq);
m_magsq = m_movingAverage.asDouble();
m_magsqSum += magsq;
if (magsq > m_magsqPeak)
{
m_magsqPeak = magsq;
}
m_magsqCount++;
// Send sample to DAB library
std::complex<float> c;
c.real(ci.real()/SDR_RX_SCALED);
c.imag(ci.imag()/SDR_RX_SCALED);
m_device.putSample(c);
}
void DABDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
{
qDebug() << "DABDemodSink::applyChannelSettings:"
<< " channelSampleRate: " << channelSampleRate
<< " channelFrequencyOffset: " << channelFrequencyOffset;
if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
(m_channelSampleRate != channelSampleRate) || force)
{
m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
}
if ((m_channelSampleRate != channelSampleRate) || force)
{
m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2);
m_interpolatorDistance = (Real) channelSampleRate / (Real) DABDEMOD_CHANNEL_SAMPLE_RATE;
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
m_channelSampleRate = channelSampleRate;
m_channelFrequencyOffset = channelFrequencyOffset;
}
void DABDemodSink::applySettings(const DABDemodSettings& settings, bool force)
{
qDebug() << "DABDemodSink::applySettings:"
<< " force: " << force;
if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
{
m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2);
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) DABDEMOD_CHANNEL_SAMPLE_RATE;
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
if ((settings.m_program != m_settings.m_program) || force)
{
if (!settings.m_program.isEmpty())
{
QByteArray ba = settings.m_program.toUtf8();
const char *program = ba.data();
if (!is_audioService (m_dab, program))
qWarning() << settings.m_program << " is not an audio service";
else
{
dataforAudioService(m_dab, program, &m_ad, 0);
if (!m_ad.defined)
qWarning() << settings.m_program << " audio data is not defined";
else
{
dabReset_msc(m_dab);
set_audioChannel(m_dab, &m_ad);
}
}
}
}
m_settings = settings;
}
// Called when audio device sample rate changes
void DABDemodSink::applyAudioSampleRate(int sampleRate)
{
if (sampleRate < 0)
{
qWarning("DABDemodSink::applyAudioSampleRate: invalid sample rate: %d", sampleRate);
return;
}
qDebug("DABDemodSink::applyAudioSampleRate: m_audioSampleRate: %d m_dabAudioSampleRate: %d", sampleRate, m_dabAudioSampleRate);
m_audioInterpolator.create(16, m_dabAudioSampleRate, m_dabAudioSampleRate/2.2f);
m_audioInterpolatorDistanceRemain = 0;
m_audioInterpolatorDistance = (Real) m_dabAudioSampleRate / (Real) sampleRate;
m_audioFifo.setSize(sampleRate);
m_audioSampleRate = sampleRate;
}
// Called when DAB audio sample rate changes
void DABDemodSink::applyDABAudioSampleRate(int sampleRate)
{
qDebug("DABDemodSink::applyDABAudioSampleRate: m_audioSampleRate: %d new m_dabAudioSampleRate: %d", m_audioSampleRate, sampleRate);
m_audioInterpolator.create(16, sampleRate, sampleRate/2.2f);
m_audioInterpolatorDistanceRemain = 0;
m_audioInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
m_dabAudioSampleRate = sampleRate;
}
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