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- adopt RFC2833 decoder for pcap analyzer

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This commit is contained in:
Dmytro Bogovych
2026-02-25 13:54:37 +03:00
parent 06b39dd629
commit 34fd9121ed
5 changed files with 116 additions and 82 deletions
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41
src/engine/media/MT_AudioReceiver.cpp
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@@ -201,7 +201,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch()
// See if there is enough information in buffer
auto total = findTimelength();
while (total > mHigh && mPacketList.size() && 0ms != mHigh)
while (total > mHigh && mPacketList.size() > 1 && 0ms != mHigh)
{
ICELogMedia( << "Dropping RTP packets from jitter buffer");
total -= mPacketList.front()->timelength();
@@ -337,7 +337,7 @@ Receiver::~Receiver()
//-------------- AudioReceiver ----------------
AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics &stat)
:Receiver(stat), mBuffer(stat), mCodecSettings(settings), mCodecList(settings)
:Receiver(stat), mBuffer(stat), mDtmfBuffer(stat), mCodecSettings(settings), mCodecList(settings), mDtmfReceiver(stat)
{
// Init resamplers
mResampler8.start(AUDIO_CHANNELS, 8000, AUDIO_SAMPLERATE);
@@ -351,6 +351,10 @@ AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics
mAvailable.setCapacity(AUDIO_SAMPLERATE * sizeof(short));
mDtmfBuffer.setPrebuffer(0ms);
mDtmfBuffer.setLow(0ms);
mDtmfBuffer.setHigh(1ms);
#if defined(DUMP_DECODED)
mDecodedDump = std::make_shared<Audio::WavFileWriter>();
mDecodedDump->open("decoded.wav", 8000 /*G711*/, AUDIO_CHANNELS);
@@ -431,14 +435,19 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
// Increase codec counter
mStat.mCodecCount[ptype]++;
// Check if we deal with telephone-event
if (p->GetPayloadType() == mCodecSettings.mTelephoneEvent)
{
*detectedCodec = nullptr;
mDtmfBuffer.add(p, 10ms, 8000);
}
else
{
// Look for codec
// Check if codec can be handled
Codec* codec = nullptr;
auto codecIter = mCodecMap.find(ptype);
if (codecIter == mCodecMap.end())
{
// Well, there is no information about the codec; skip this packet
}
else
if (codecIter != mCodecMap.end())
{
// Check if codec is creating lazily
if (!codecIter->second)
@@ -467,7 +476,7 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
samplerate = codec->samplerate();
}
// Process jitter
// Process jitter anyway - can we decode payload or not
mJitterStats.process(p.get(), samplerate);
mStat.mJitter = static_cast<float>(mJitterStats.get());
@@ -489,9 +498,10 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
// Queue packet to buffer
auto packet = mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate).get();
return packet;
}
return {};
}
void AudioReceiver::processDecoded(Audio::DataWindow& output, DecodeOptions options)
{
@@ -764,6 +774,12 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
{
DecodeResult result = {.mStatus = DecodeResult::Status::Skip};
// Process RFC2833 here; it doesn't result in any audio - only callbacks and statistics
auto fr = mDtmfBuffer.fetch();
if (fr.mPacket && fr.mStatus == RtpBuffer::FetchResult::Status::RegularPacket)
mDtmfReceiver.add(fr.mPacket->rtp());
auto produced = 0ms;
if (mAvailable.filled() && mCodec && options.mElapsed != 0ms)
{
@@ -968,12 +984,10 @@ DtmfReceiver::~DtmfReceiver()
void DtmfReceiver::add(const std::shared_ptr<RTPPacket>& p)
{
// This receiver always work in context of single RTP stream; so there is no need to put SSRC map and so on
if (p->GetPayloadType() != 101)
return;
auto ev = DtmfBuilder::parseRfc2833({p->GetPayloadData(), p->GetPayloadLength()});
if (ev.mTone != mEvent || ev.mEnd != mEventEnded)
{
if (!(mEvent == ev.mTone && !mEventEnded && ev.mEnd))
{
// New tone is here
if (mCallback)
@@ -988,3 +1002,4 @@ void DtmfReceiver::add(const std::shared_ptr<RTPPacket>& p)
mEventEnded = ev.mEnd;
}
}
}

35
src/engine/media/MT_AudioReceiver.h
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@@ -136,6 +136,23 @@ protected:
Statistics& mStat;
};
class DtmfReceiver: public Receiver
{
private:
char mEvent = 0;
bool mEventEnded = false;
std::chrono::milliseconds mEventStart = 0ms;
std::function<void(char)> mCallback;
public:
DtmfReceiver(Statistics& stat);
~DtmfReceiver();
void add(const std::shared_ptr<RTPPacket>& p);
void setCallback(std::function<void(char tone)> callback);
};
class AudioReceiver: public Receiver
{
public:
@@ -189,6 +206,9 @@ public:
protected:
RtpBuffer mBuffer; // Jitter buffer itself
RtpBuffer mDtmfBuffer; // These two (mDtmfBuffer / mDtmfReceiver) are for our analyzer stack only; in normal softphone logic DTMF packets goes via SingleAudioStream::mDtmfReceiver
DtmfReceiver mDtmfReceiver;
CodecMap mCodecMap;
PCodec mCodec;
int mFrameCount = 0;
@@ -247,21 +267,6 @@ protected:
DecodeResult decodeEmptyTo(Audio::DataWindow& output, DecodeOptions options);
};
class DtmfReceiver: public Receiver
{
private:
char mEvent = 0;
bool mEventEnded = false;
std::chrono::milliseconds mEventStart = 0ms;
std::function<void(char)> mCallback;
public:
DtmfReceiver(Statistics& stat);
~DtmfReceiver();
void add(const std::shared_ptr<RTPPacket>& p);
void setCallback(std::function<void(char tone)> callback);
};
}
#endif

17
src/engine/media/MT_CodecList.cpp
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@@ -65,6 +65,8 @@ bool CodecList::Settings::contains(int ptype) const
if (mGsmEfrPayloadType == ptype || mGsmFrPayloadType == ptype || mGsmHrPayloadType == ptype)
return true;
if (mTelephoneEvent == ptype)
return true;
return false;
}
@@ -122,6 +124,9 @@ std::string CodecList::Settings::toString() const
if (mGsmEfrPayloadType != -1)
oss << "GSM EFR ptype: " << mGsmEfrPayloadType << " ";
if (mTelephoneEvent != -1)
oss << "RFC2833 DTMF ptype: " << mTelephoneEvent;
for (auto& spec: mEvsSpec)
{
oss << "EVS ptype: " << spec.mPayloadType << ", bw: " << spec.mBandwidth << ", enc: " << (spec.mEncodingType == EvsSpec::Encoding_MIME ? "mime" : "g192") << " ";
@@ -132,6 +137,7 @@ std::string CodecList::Settings::toString() const
oss << "OPUS ptype: " << spec.mPayloadType << ", rate: " << spec.mRate << ", channels: " << spec.mChannels << std::endl;
}
return oss.str();
}
@@ -151,6 +157,7 @@ void CodecList::Settings::clear()
mGsmEfrPayloadType = -1;
mGsmFrPayloadType = -1;
mGsmHrPayloadType = -1;
mTelephoneEvent = -1;
}
bool CodecList::Settings::EvsSpec::isValid() const
@@ -268,15 +275,16 @@ CodecList::Settings CodecList::Settings::parseSdp(const std::list<resip::Codec>&
}
} else if (codec_name == "EVS") {
r.mEvsSpec.push_back({ptype});
}
} else if (codec_name == "TELEPHONE-EVENT")
r.mTelephoneEvent = ptype;
}
return r;
}
bool CodecList::Settings::operator == (const Settings& rhs) const
{
if (std::tie(mWrapIuUP, mSkipDecode, mIsac16KPayloadType, mIsac32KPayloadType, mIlbc20PayloadType, mIlbc30PayloadType, mGsmFrPayloadType, mGsmFrPayloadLength, mGsmEfrPayloadType, mGsmHrPayloadType) !=
std::tie(rhs.mWrapIuUP, rhs.mSkipDecode, rhs.mIsac16KPayloadType, rhs.mIsac32KPayloadType, rhs.mIlbc20PayloadType, rhs.mIlbc30PayloadType, rhs.mGsmFrPayloadType, rhs.mGsmFrPayloadLength, rhs.mGsmEfrPayloadType, rhs.mGsmHrPayloadType))
if (std::tie(mWrapIuUP, mSkipDecode, mIsac16KPayloadType, mIsac32KPayloadType, mIlbc20PayloadType, mIlbc30PayloadType, mGsmFrPayloadType, mGsmFrPayloadLength, mGsmEfrPayloadType, mGsmHrPayloadType, mTelephoneEvent) !=
std::tie(rhs.mWrapIuUP, rhs.mSkipDecode, rhs.mIsac16KPayloadType, rhs.mIsac32KPayloadType, rhs.mIlbc20PayloadType, rhs.mIlbc30PayloadType, rhs.mGsmFrPayloadType, rhs.mGsmFrPayloadLength, rhs.mGsmEfrPayloadType, rhs.mGsmHrPayloadType, rhs.mTelephoneEvent))
return false;
if (mAmrNbOctetPayloadType != rhs.mAmrNbOctetPayloadType)
@@ -306,6 +314,9 @@ bool CodecList::Settings::operator == (const Settings& rhs) const
if (mOpusSpec[i] != rhs.mOpusSpec[i])
return false;
if (mTelephoneEvent != rhs.mTelephoneEvent)
return false;
return true;
}

3
src/engine/media/MT_CodecList.h
View File

@@ -28,6 +28,9 @@ public:
bool mWrapIuUP = false;
bool mSkipDecode = false;
// RFC2833 DTMF
int mTelephoneEvent = -1;
// AMR payload types
std::set<int64_t> mAmrWbPayloadType = { };
std::set<int64_t> mAmrNbPayloadType = { };

2
src/engine/media/MT_SingleAudioStream.cpp
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@@ -23,7 +23,7 @@ SingleAudioStream::~SingleAudioStream()
void SingleAudioStream::process(const std::shared_ptr<jrtplib::RTPPacket>& packet)
{
ICELogMedia(<< "Processing incoming RTP/RTCP packet");
if (packet->GetPayloadType() == 101/*resip::Codec::TelephoneEvent.payloadType()*/)
if (packet->GetPayloadType() == mReceiver.getCodecSettings().mTelephoneEvent)
mDtmfReceiver.add(packet);
else
mReceiver.add(packet);