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RTP audio sink fixes and test in NFM demod (endianess problem)

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This commit is contained in:
f4exb 2018-01-31 00:40:54 +01:00
parent f44df73590
commit 70adea206c
7 changed files with 289 additions and 143 deletions
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6
plugins/channelrx/demodnfm/nfmdemod.cpp
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@ -83,6 +83,12 @@ NFMDemod::NFMDemod(DeviceSourceAPI *devieAPI) :
m_audioNetSink = new AudioNetSink(this);
m_audioNetSink->setDestination(m_settings.m_udpAddress, m_settings.m_udpPort);
if (m_audioNetSink->selectType(AudioNetSink::SinkRTP)) {
qDebug("NFMDemod::NFMDemod: set audio sink to RTP mode");
} else {
qWarning("NFMDemod::NFMDemod: RTP support for audio sink not available. Fall back too UDP");
}
m_channelizer = new DownChannelizer(this);
m_threadedChannelizer = new ThreadedBasebandSampleSink(m_channelizer, this);
m_deviceAPI->addThreadedSink(m_threadedChannelizer);

2
plugins/channelrx/demodnfm/nfmdemodsettings.cpp
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@ -52,7 +52,7 @@ void NFMDemodSettings::resetToDefaults()
m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
m_copyAudioToUDP = false;
m_udpAddress = "127.0.0.1";
m_udpPort = 9999;
m_udpPort = 9998;
m_rgbColor = QColor(255, 0, 0).rgb();
m_title = "NFM Demodulator";
}

4
sdrbase/CMakeLists.txt
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@ -216,6 +216,10 @@ if (JRTPLIB_FOUND)
${sdrbase_HEADERS}
util/rtpsink.h
)
set(sdrbase_SOURCES
${sdrbase_SOURCES}
util/rtpsink.cpp
)
add_definitions(-DHAS_JRTPLIB)
include_directories(${JRTPLIB_INCLUDE_DIR})
endif(JRTPLIB_FOUND)

84
sdrbase/audio/audionetsink.cpp
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@ -26,33 +26,33 @@ const int AudioNetSink::m_udpBlockSize = 512;
AudioNetSink::AudioNetSink(QObject *parent, bool stereo) :
m_type(SinkUDP),
m_udpBufferAudioMono(0),
m_udpBufferAudioStereo(0)
m_udpBufferAudioStereo(0),
m_rtpBufferAudio(0)
{
if (stereo)
{
if (stereo) {
m_udpBufferAudioStereo = new UDPSink<AudioSample>(parent, m_udpBlockSize);
#ifdef HAS_JRTPLIB
m_rtpBufferAudioStereo = new RTPSink<AudioSample>("127.0.0.1", 9999, 48000);
m_rtpBufferAudioMono = 0;
#endif
}
else
{
} else {
m_udpBufferAudioMono = new UDPSink<int16_t>(parent, m_udpBlockSize);
#ifdef HAS_JRTPLIB
m_rtpBufferAudioMono = new RTPSink<int16_t>("127.0.0.1", 9999, 48000);
m_rtpBufferAudioStereo = 0;
#endif
}
#ifdef HAS_JRTPLIB
m_rtpBufferAudio = new RTPSink("127.0.0.1", 9999, stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono);
#endif
}
AudioNetSink::~AudioNetSink()
{
if (m_udpBufferAudioMono) { delete m_udpBufferAudioMono; }
if (m_udpBufferAudioStereo) { delete m_udpBufferAudioStereo; }
if (m_udpBufferAudioMono) {
delete m_udpBufferAudioMono;
}
if (m_udpBufferAudioStereo) {
delete m_udpBufferAudioStereo;
}
#ifdef HAS_JRTPLIB
if (m_rtpBufferAudioMono) { delete m_rtpBufferAudioMono; }
if (m_rtpBufferAudioStereo) { delete m_rtpBufferAudioStereo; }
if (m_rtpBufferAudio) {
delete m_rtpBufferAudio;
}
#endif
}
@ -87,51 +87,39 @@ void AudioNetSink::setDestination(const QString& address, uint16_t port)
if (m_udpBufferAudioStereo) {
m_udpBufferAudioStereo->setDestination(address, port);
}
#ifdef HAS_JRTPLIB
if (m_rtpBufferAudioMono) {
m_rtpBufferAudioMono->setDestination(address, port);
}
if (m_rtpBufferAudioStereo) {
m_rtpBufferAudioStereo->setDestination(address, port);
if (m_rtpBufferAudio) {
m_rtpBufferAudio->setDestination(address, port);
}
#endif
}
#ifdef HAS_JRTPLIB
void AudioNetSink::addDestination(const QString& address, uint16_t port)
{
#ifdef HAS_JRTPLIB
if (m_rtpBufferAudioMono) {
m_rtpBufferAudioMono->addDestination(address, port);
if (m_rtpBufferAudio) {
m_rtpBufferAudio->addDestination(address, port);
}
if (m_rtpBufferAudioStereo) {
m_rtpBufferAudioStereo->addDestination(address, port);
}
#endif
}
#else
void AudioNetSink::addDestination(const QString& address __attribute__((unused)), uint16_t port __attribute__((unused)))
{
}
#endif
#ifdef HAS_JRTPLIB
void AudioNetSink::deleteDestination(const QString& address, uint16_t port)
{
#ifdef HAS_JRTPLIB
if (m_rtpBufferAudioMono) {
m_rtpBufferAudioMono->deleteDestination(address, port);
if (m_rtpBufferAudio) {
m_rtpBufferAudio->deleteDestination(address, port);
}
if (m_rtpBufferAudioStereo) {
m_rtpBufferAudioStereo->deleteDestination(address, port);
}
#endif
}
void AudioNetSink::setSampleRate(int sampleRate)
#else
void AudioNetSink::deleteDestination(const QString& address __attribute__((unused)), uint16_t port __attribute__((unused)))
{
#ifdef HAS_JRTPLIB
if (m_rtpBufferAudioMono) {
m_rtpBufferAudioMono->setSampleRate(sampleRate);
}
if (m_rtpBufferAudioStereo) {
m_rtpBufferAudioStereo->setSampleRate(sampleRate);
}
#endif
}
#endif
void AudioNetSink::write(qint16 sample)
{
@ -143,6 +131,7 @@ void AudioNetSink::write(qint16 sample)
m_udpBufferAudioMono->write(sample);
} else if (m_type == SinkRTP) {
#ifdef HAS_JRTPLIB
m_rtpBufferAudio->write((uint8_t *) &sample);
#endif
}
}
@ -157,6 +146,7 @@ void AudioNetSink::write(const AudioSample& sample)
m_udpBufferAudioStereo->write(sample);
} else if (m_type == SinkRTP) {
#ifdef HAS_JRTPLIB
m_rtpBufferAudio->write((uint8_t *) &sample);
#endif
}
}

6
sdrbase/audio/audionetsink.h
View File

@ -23,7 +23,7 @@
#include "util/export.h"
template<typename T> class UDPSink;
template<typename T> class RTPSink;
class RTPSink;
class SDRANGEL_API AudioNetSink {
public:
@ -39,7 +39,6 @@ public:
void setDestination(const QString& address, uint16_t port);
void addDestination(const QString& address, uint16_t port);
void deleteDestination(const QString& address, uint16_t port);
void setSampleRate(int sampleRate);
void write(qint16 sample);
void write(const AudioSample& sample);
@ -52,8 +51,7 @@ protected:
SinkType m_type;
UDPSink<qint16> *m_udpBufferAudioMono;
UDPSink<AudioSample> *m_udpBufferAudioStereo;
RTPSink<qint16> *m_rtpBufferAudioMono;
RTPSink<AudioSample> *m_rtpBufferAudioStereo;
RTPSink *m_rtpBufferAudio;
};

220
sdrbase/util/rtpsink.cpp Normal file
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@ -0,0 +1,220 @@
///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2018 F4EXB //
// written by Edouard Griffiths //
// //
// 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 //
// //
// 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 "rtpsink.h"
#include "dsp/dsptypes.h"
RTPSink::RTPSink(const QString& address, uint16_t port, PayloadType payloadType) :
m_sampleRate(48000),
m_sampleBytes(0),
m_packetSamples(0),
m_bufferSize(0),
m_sampleBufferIndex(0),
m_byteBuffer(0),
m_destport(port),
m_mutex(QMutex::Recursive)
{
qDebug("RTPSink::RTPSink");
m_destip = inet_addr(address.toStdString().c_str());
m_destip = ntohl(m_destip);
m_rtpSessionParams.SetOwnTimestampUnit(1.0 / (double) m_sampleRate);
int status = m_rtpSession.Create(m_rtpSessionParams);
if (status < 0) {
qCritical("RTPSink::RTPSink: cannot create session: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::RTPSink: created session: %s", jrtplib::RTPGetErrorString(status).c_str());
}
setPayloadType(payloadType);
}
RTPSink::~RTPSink()
{
jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
m_rtpSession.BYEDestroy(delay, "Time's up", 9);
if (m_byteBuffer) {
delete[] m_byteBuffer;
}
}
void RTPSink::setPayloadType(PayloadType payloadType)
{
QMutexLocker locker(&m_mutex);
qDebug("RTPSink::setPayloadType: %d", payloadType);
switch (payloadType)
{
case PayloadL16Stereo:
m_sampleRate = 48000;
m_sampleBytes = sizeof(AudioSample);
m_rtpSession.SetDefaultPayloadType(96);
break;
case PayloadL16Mono:
default:
m_sampleRate = 48000;
m_sampleBytes = sizeof(int16_t);
m_rtpSession.SetDefaultPayloadType(96);
break;
}
m_packetSamples = m_sampleRate/50; // 20ms packet samples
m_bufferSize = m_packetSamples * m_sampleBytes;
if (m_byteBuffer) {
delete[] m_byteBuffer;
}
m_byteBuffer = new uint8_t[m_bufferSize];
m_sampleBufferIndex = 0;
m_payloadType = payloadType;
int status = m_rtpSession.SetTimestampUnit(1.0 / (double) m_sampleRate);
if (status < 0) {
qCritical("RTPSink::setPayloadType: cannot set timestamp unit: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::setPayloadType: timestamp unit set to %f: %s",
1.0 / (double) m_sampleRate,
jrtplib::RTPGetErrorString(status).c_str());
}
status = m_rtpSession.SetDefaultMark(false);
if (status < 0) {
qCritical("RTPSink::setPayloadType: cannot set default mark: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::setPayloadType: set default mark to false: %s", jrtplib::RTPGetErrorString(status).c_str());
}
status = m_rtpSession.SetDefaultTimestampIncrement(m_packetSamples);
if (status < 0) {
qCritical("RTPSink::setPayloadType: cannot set default timestamp increment: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::setPayloadType: set default timestamp increment to %d: %s", m_packetSamples, jrtplib::RTPGetErrorString(status).c_str());
}
status = m_rtpSession.SetMaximumPacketSize(m_bufferSize+40);
if (status < 0) {
qCritical("RTPSink::setPayloadType: cannot set maximum packet size: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::setPayloadType: set maximum packet size to %d bytes: %s", m_bufferSize+40, jrtplib::RTPGetErrorString(status).c_str());
}
}
void RTPSink::setDestination(const QString& address, uint16_t port)
{
m_rtpSession.ClearDestinations();
m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
m_destip = inet_addr(address.toStdString().c_str());
m_destip = ntohl(m_destip);
m_destport = port;
int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
if (status < 0) {
qCritical("RTPSink::setDestination: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
}
}
void RTPSink::deleteDestination(const QString& address, uint16_t port)
{
uint32_t destip = inet_addr(address.toStdString().c_str());
destip = ntohl(m_destip);
int status = m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(destip, port));
if (status < 0) {
qCritical("RTPSink::deleteDestination: cannot delete destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
}
}
void RTPSink::addDestination(const QString& address, uint16_t port)
{
uint32_t destip = inet_addr(address.toStdString().c_str());
destip = ntohl(m_destip);
int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(destip, port));
if (status < 0) {
qCritical("RTPSink::addDestination: cannot add destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
} else {
qDebug("RTPSink::addDestination: destination address set to %s:%d: %s",
address.toStdString().c_str(),
port,
jrtplib::RTPGetErrorString(status).c_str());
}
}
void RTPSink::write(uint8_t *sampleByte)
{
QMutexLocker locker(&m_mutex);
if (m_sampleBufferIndex < m_packetSamples)
{
memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes], sampleByte, m_sampleBytes);
m_sampleBufferIndex++;
}
else
{
int status = m_rtpSession.SendPacket((const void *) m_byteBuffer, (std::size_t) m_bufferSize);
if (status < 0) {
qCritical("RTPSink::write: cannot write packet: %s", jrtplib::RTPGetErrorString(status).c_str());
}
memcpy(&m_byteBuffer[0], sampleByte, m_sampleBytes);
m_sampleBufferIndex = 1;
}
}
void RTPSink::write(uint8_t *samples, int nbSamples)
{
int samplesIndex = 0;
QMutexLocker locker(&m_mutex);
// fill remainder of buffer and send it
if (m_sampleBufferIndex + nbSamples > m_packetSamples)
{
memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes],
&samples[samplesIndex*m_sampleBytes],
(m_packetSamples - m_sampleBufferIndex)*m_sampleBytes);
m_rtpSession.SendPacket((const void *) m_byteBuffer, (std::size_t) m_bufferSize);
nbSamples -= (m_packetSamples - m_sampleBufferIndex);
m_sampleBufferIndex = 0;
}
// send complete packets
while (nbSamples > m_packetSamples)
{
m_rtpSession.SendPacket((const void *) &samples[samplesIndex*m_sampleBytes], (std::size_t) m_bufferSize);
samplesIndex += m_packetSamples;
nbSamples -= m_packetSamples;
}
// copy remainder of input to buffer
memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes],
&samples[samplesIndex*m_sampleBytes],
nbSamples*m_sampleBytes);
}

110
sdrbase/util/rtpsink.h
View File

@ -30,112 +30,40 @@
#include "rtperrors.h"
#include "rtplibraryversion.h"
template<typename SampleType>
class RTPSink
{
public:
RTPSink(const QString& address, uint16_t port, int sampleRate) :
m_destport(port),
m_sampleBuffer(0),
m_sampleBufferIndex(0),
m_mutex(QMutex::Recursive)
typedef enum
{
m_destip = inet_addr(address.toStdString().c_str());
m_destip = ntohl(m_destip);
m_rtpSessionParams.SetOwnTimestampUnit(1.0/(double) sampleRate);
m_rtpTransmissionParams.SetPortbase(8092); // FIXME: sort this out
PayloadL16Mono,
PayloadL16Stereo,
} PayloadType;
int status = m_rtpSession.Create(m_rtpSessionParams, &m_rtpTransmissionParams);
RTPSink(const QString& address, uint16_t port, PayloadType payloadType = PayloadL16Mono);
~RTPSink();
if (status < 0)
{
qCritical("RTPSink::RTPSink: cannot create session: %s", jrtplib::RTPGetErrorString(status).c_str());
return;
}
void setPayloadType(PayloadType payloadType);
status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
void setDestination(const QString& address, uint16_t port);
void deleteDestination(const QString& address, uint16_t port);
void addDestination(const QString& address, uint16_t port);
if (status < 0)
{
qCritical("RTPSink::RTPSink: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
return;
}
m_sampleBuffer = new SampleType[sampleRate]; // store 1 second
}
~RTPSink()
{
jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
m_rtpSession.BYEDestroy(delay, "Time's up", 9);
if (m_sampleBuffer) { delete[] m_sampleBuffer; }
}
void setDestination(const QString& address, uint16_t port)
{
if (!m_sampleBuffer) { return; }
m_rtpSession.ClearDestinations();
m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
m_destip = inet_addr(address.toStdString().c_str());
m_destip = ntohl(m_destip);
m_destport = port;
int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
if (status < 0) {
qCritical("RTPSink::setDestination: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
}
}
void deleteDestination(const QString& address, uint16_t port)
{
uint32_t destip = inet_addr(address.toStdString().c_str());
destip = ntohl(m_destip);
int status = m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(destip, port));
if (status < 0) {
qCritical("RTPSink::deleteDestination: cannot delete destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
}
}
void addDestination(const QString& address, uint16_t port)
{
uint32_t destip = inet_addr(address.toStdString().c_str());
destip = ntohl(m_destip);
int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(destip, port));
if (status < 0) {
qCritical("RTPSink::addDestination: cannot add destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
}
}
void setSampleRate(int sampleRate)
{
QMutexLocker locker(&m_mutex);
if (m_sampleBuffer) { delete[] m_sampleBuffer; }
m_sampleBuffer = new SampleType[sampleRate]; // store 1 second
int status = m_rtpSession.SetTimestampUnit(1.0 / (double) sampleRate);
if (status < 0)
{
qCritical("RTPSink::setSampleRate: cannot set timestamp unit: %s", jrtplib::RTPGetErrorString(status).c_str());
return;
}
}
void write(uint8_t *sampleByte);
void write(uint8_t *sampleByte, int nbSamples);
protected:
PayloadType m_payloadType;
int m_sampleRate;
int m_sampleBytes;
int m_packetSamples;
int m_bufferSize;
int m_sampleBufferIndex;
uint8_t *m_byteBuffer;
uint32_t m_destip;
uint16_t m_destport;
jrtplib::RTPSession m_rtpSession;
jrtplib::RTPSessionParams m_rtpSessionParams;
jrtplib::RTPUDPv4TransmissionParams m_rtpTransmissionParams;
SampleType *m_sampleBuffer;
int m_sampleBufferIndex;
QMutex m_mutex;
};