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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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src/engine/media/MT_AudioReceiver.cpp
+78 -63
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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);
@@ -431,66 +435,72 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
// Increase codec counter
mStat.mCodecCount[ptype]++;
// Check if codec can be handled
Codec* codec = nullptr;
auto codecIter = mCodecMap.find(ptype);
if (codecIter == mCodecMap.end())
// Check if we deal with telephone-event
if (p->GetPayloadType() == mCodecSettings.mTelephoneEvent)
{
// Well, there is no information about the codec; skip this packet
*detectedCodec = nullptr;
mDtmfBuffer.add(p, 10ms, 8000);
}
else
{
// Check if codec is creating lazily
if (!codecIter->second)
// Look for codec
// Check if codec can be handled
Codec* codec = nullptr;
auto codecIter = mCodecMap.find(ptype);
if (codecIter != mCodecMap.end())
{
codecIter->second = mCodecList.createCodecByPayloadType(ptype);
// Check if codec is creating lazily
if (!codecIter->second)
{
codecIter->second = mCodecList.createCodecByPayloadType(ptype);
}
codec = codecIter->second.get();
// Return pointer to codec if needed.get()
if (detectedCodec)
*detectedCodec = codec;
if (mStat.mCodecName.empty() && codec)
mStat.mCodecName = codec->name();
if (!codec)
time_length = 10;
else
if (!codec->rtpLength())
time_length = codec->frameTime();
else
time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
if (codec)
samplerate = codec->samplerate();
}
codec = codecIter->second.get();
// Return pointer to codec if needed.get()
if (detectedCodec)
*detectedCodec = codec;
if (mStat.mCodecName.empty() && codec)
mStat.mCodecName = codec->name();
// Process jitter anyway - can we decode payload or not
mJitterStats.process(p.get(), samplerate);
mStat.mJitter = static_cast<float>(mJitterStats.get());
if (!codec)
time_length = 10;
return false; // There is no sense to add this packet into jitter buffer - we can't decode this
// Check if packet is CNG
if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
else
if (!codec->rtpLength())
time_length = codec->frameTime();
else
time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
// Check if packet is too short from time length side
if (time_length < 2)
{
// It will cause statistics to report about bad RTP packet
// I have to replay last packet payload here to avoid report about lost packet
mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
return false;
}
if (codec)
samplerate = codec->samplerate();
// Queue packet to buffer
auto packet = mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate).get();
return packet;
}
// Process jitter
mJitterStats.process(p.get(), samplerate);
mStat.mJitter = static_cast<float>(mJitterStats.get());
if (!codec)
return false; // There is no sense to add this packet into jitter buffer - we can't decode this
// Check if packet is CNG
if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
else
// Check if packet is too short from time length side
if (time_length < 2)
{
// It will cause statistics to report about bad RTP packet
// I have to replay last packet payload here to avoid report about lost packet
mBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
return false;
}
// 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)
{
@@ -968,23 +984,22 @@ 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)
{
// New tone is here
if (mCallback)
mCallback(ev.mTone);
if (!(mEvent == ev.mTone && !mEventEnded && ev.mEnd))
{
// New tone is here
if (mCallback)
mCallback(ev.mTone);
// Queue statistics item
mStat.mDtmf2833Timeline.emplace_back(Dtmf2833Event{.mTone = ev.mTone,
.mTimestamp = RtpHelper::toMicroseconds(p->GetReceiveTime())});
// Queue statistics item
mStat.mDtmf2833Timeline.emplace_back(Dtmf2833Event{.mTone = ev.mTone,
.mTimestamp = RtpHelper::toMicroseconds(p->GetReceiveTime())});
// Store to avoid triggering on the packet
mEvent = ev.mTone;
mEventEnded = ev.mEnd;
// Store to avoid triggering on the packet
mEvent = ev.mTone;
mEventEnded = ev.mEnd;
}
}
}