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- source codes cleanups and minor fixes

This commit is contained in:
Dmytro Bogovych 2025-04-16 09:58:06 +03:00
parent a0b7cdcd99
commit 8d0c8ba4de
4 changed files with 182 additions and 173 deletions
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76
src/engine/media/MT_AudioReceiver.cpp
View File

@ -62,6 +62,8 @@ std::vector<short>& RtpBuffer::Packet::pcm()
RtpBuffer::RtpBuffer(Statistics& stat)
:mStat(stat)
{
if (mStat.mPacketLoss)
std::cout << "Warning: packet loss is not zero" << std::endl;
}
RtpBuffer::~RtpBuffer()
@ -203,6 +205,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
// Save it as last packet however - to not confuse loss packet counter
mFetchedPacket = mPacketList.front();
mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
// Erase from packet list
mPacketList.erase(mPacketList.begin());
@ -212,47 +215,48 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
}
if (total < mLow)
{
// Still not prebuffered
result = FetchResult::NoPacket;
}
else
{
// Did we fetch any packet before ?
bool is_fetched_packet = mFetchedPacket.get() != nullptr;
if (is_fetched_packet)
is_fetched_packet &= mFetchedPacket->rtp().get() != nullptr;
if (is_fetched_packet)
if (mLastSeqno.has_value())
{
if (mPacketList.empty())
{
result = FetchResult::NoPacket;
mStat.mPacketLoss++;
// Don't increase counter of lost packets here; maybe it is DTX
}
else
{
// Current sequence number ?
unsigned seqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
uint32_t seqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
// Gap between new packet and previous on
int gap = (int64_t)seqno - (int64_t)mFetchedPacket->rtp()->GetExtendedSequenceNumber() - 1;
int gap = (int64_t)seqno - (int64_t)*mLastSeqno - 1;
gap = std::min(gap, 127);
if (gap > 0 && mPacketList.empty())
if (gap > 0)
{
// std::cout << "Increase the packet loss for SSRC " << std::hex << mSsrc << std::endl;
mStat.mPacketLoss++;
//mStat.mLoss[gap]++;
mLastSeqno = *mLastSeqno + 1;
result = FetchResult::Gap;
mStat.mPacketLoss += gap;
mStat.mLoss[gap]++;
}
else
{
if (gap > 0)
{
mStat.mPacketLoss += gap;
mStat.mLoss[gap]++;
}
result = FetchResult::RegularPacket;
rl.push_back(mPacketList.front());
// Save last returned normal packet
mFetchedPacket = mPacketList.front();
mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
// Remove returned packet from the list
mPacketList.erase(mPacketList.begin());
@ -262,7 +266,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
else
{
// See if prebuffer limit is reached
if (findTimelength() >= mPrebuffer)
if (findTimelength() >= mPrebuffer && !mPacketList.empty())
{
// Normal packet will be returned
result = FetchResult::RegularPacket;
@ -272,6 +276,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
// Remember returned packet
mFetchedPacket = mPacketList.front();
mLastSeqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
// Remove returned packet from buffer list
mPacketList.erase(mPacketList.begin());
@ -402,10 +407,11 @@ size_t decode_packet(Codec& codec, RTPPacket& p, void* output_buffer, size_t out
return result;
}
bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec)
bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** detectedCodec)
{
// Estimate time length
int time_length = 0,
samplerate = 8000,
payloadLength = p->GetPayloadLength(),
ptype = p->GetPayloadType();
@ -414,34 +420,43 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** co
mStat.mCodecCount[ptype]++;
// Check if codec can be handled
Codec* codec = nullptr;
CodecMap::iterator codecIter = mCodecMap.find(ptype);
if (codecIter == mCodecMap.end())
{
ICELogMedia(<< "Cannot find codec in available codecs");
return false; // Reject packet with unknown payload type
time_length = 10;
}
else
{
// 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 (codec)
*codec = codecIter->second.get();
if (detectedCodec)
*detectedCodec = codec;
if (mStat.mCodecName.empty())
mStat.mCodecName = codecIter->second->name();
if (mStat.mCodecName.empty() && codec)
mStat.mCodecName = codec->name();
if (!codecIter->second->rtpLength())
time_length = codecIter->second->frameTime();
if (!codec)
time_length = 10;
else
time_length = lround(double(payloadLength) / codecIter->second->rtpLength() * codecIter->second->frameTime());
if (!codec->rtpLength())
time_length = codec->frameTime();
else
time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
if (codec)
samplerate = codec->samplerate();
}
// Process jitter
mJitterStats.process(p.get(), codecIter->second->samplerate());
mJitterStats.process(p.get(), samplerate);
mStat.mJitter = static_cast<float>(mJitterStats.get());
// Check if packet is CNG
@ -453,20 +468,20 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** co
{
// 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, time_length, codecIter->second->samplerate());
mBuffer.add(p, time_length, samplerate);
return false;
}
// Queue packet to buffer
auto packet = mBuffer.add(p, time_length, codecIter->second->samplerate()).get();
auto packet = mBuffer.add(p, time_length, samplerate).get();
if (packet)
{
// Check if early decoding configured
if (mEarlyDecode && *codec)
if (mEarlyDecode && codec)
{
// Move data to packet buffer
size_t available = decode_packet(**codec, *p, mDecodedFrame, sizeof mDecodedFrame);
size_t available = decode_packet(*codec, *p, mDecodedFrame, sizeof mDecodedFrame);
if (available > 0)
{
packet->pcm().resize(available / 2);
@ -549,7 +564,6 @@ AudioReceiver::DecodeResult AudioReceiver::getAudio(Audio::DataWindow& output, i
case RtpBuffer::FetchResult::RegularPacket:
mFailedCount = 0;
for (std::shared_ptr<RtpBuffer::Packet>& p: rl)
{
assert(p);

11
src/engine/media/MT_AudioReceiver.h
View File

@ -1,4 +1,4 @@
/* Copyright(C) 2007-2017 VoIPobjects (voipobjects.com)
/* Copyright(C) 2007-2025 VoIPobjects (voipobjects.com)
* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
@ -8,22 +8,16 @@
#include "MT_Stream.h"
#include "MT_CodecList.h"
#include "MT_AudioCodec.h"
#include "MT_CngHelper.h"
#include "../helper/HL_Pointer.h"
#include "../helper/HL_Sync.h"
#include "../helper/HL_Optional.hpp"
#include "jrtplib/src/rtppacket.h"
#include "jrtplib/src/rtcppacket.h"
#include "jrtplib/src/rtpsourcedata.h"
#include "../audio/Audio_DataWindow.h"
#include "../audio/Audio_Resampler.h"
#include <map>
// #define DUMP_DECODED
#include <optional>
namespace MT
{
@ -101,6 +95,7 @@ namespace MT
bool mFirstPacketWillGo = true;
jrtplib::RTPSourceStats mRtpStats;
std::shared_ptr<Packet> mFetchedPacket;
std::optional<uint32_t> mLastSeqno;
// To calculate average interval between packet add. It is close to jitter but more useful in debugging.
float mLastAddTime = 0.0;

2
src/engine/media/MT_CodecList.cpp
View File

@ -423,7 +423,7 @@ PCodec CodecList::createCodecByPayloadType(int payloadType)
if (factory->payloadType() == payloadType)
return factory->create();
}
return PCodec();
return {};
}
CodecListPriority::CodecListPriority()

2
src/libs/ice/ICEAddress.cpp
View File

@ -68,7 +68,7 @@ NetworkAddress NetworkAddress::parse(const std::string& s)
if (ip4Pos == std::string::npos && ip6Pos == std::string::npos)
{
// Parse usual IP[:port] pair
std::string::size_type cp = s.find(":");
std::string::size_type cp = s.find_last_of(":");
if (cp == std::string::npos)
result.setIp(cp);
else