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Merge remote-tracking branch 'origin/stable'

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This commit is contained in:
Dmytro Bogovych
2022-06-03 08:46:24 +03:00
parent 2fa4d0c3e6 c3c59ddf03
commit 93d53957a2
1332 changed files with 134626 additions and 292 deletions
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299
src/engine/media/MT_AudioReceiver.cpp
View File

@@ -10,6 +10,7 @@
#include "MT_AudioCodec.h"
#include "MT_CngHelper.h"
#include "../helper/HL_Log.h"
#include "../helper/HL_Time.h"
#include "../audio/Audio_Interface.h"
#include "../audio/Audio_Resampler.h"
#include <cmath>
@@ -47,11 +48,19 @@ int RtpBuffer::Packet::rate() const
return mRate;
}
const std::vector<short>& RtpBuffer::Packet::pcm() const
{
return mPcm;
}
std::vector<short>& RtpBuffer::Packet::pcm()
{
return mPcm;
}
// ------------ RtpBuffer ----------------
RtpBuffer::RtpBuffer(Statistics& stat)
:mStat(stat), mSsrc(0), mHigh(RTP_BUFFER_HIGH), mLow(RTP_BUFFER_LOW), mPrebuffer(RTP_BUFFER_PREBUFFER),
mFirstPacketWillGo(true), mReturnedCounter(0), mAddCounter(0),
mFetchedPacket(std::shared_ptr<RTPPacket>(), 0, 0)
:mStat(stat)
{
}
@@ -96,19 +105,28 @@ int RtpBuffer::getCount() const
return static_cast<int>(mPacketList.size());
}
bool SequenceSort(const RtpBuffer::Packet& p1, const RtpBuffer::Packet& p2)
bool SequenceSort(const std::shared_ptr<RtpBuffer::Packet>& p1, const std::shared_ptr<RtpBuffer::Packet>& p2)
{
return p1.rtp()->GetExtendedSequenceNumber() < p2.rtp()->GetExtendedSequenceNumber();
return p1->rtp()->GetExtendedSequenceNumber() < p2->rtp()->GetExtendedSequenceNumber();
}
bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength, int rate)
std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength, int rate)
{
if (!packet)
return false;
return std::shared_ptr<Packet>();
Lock l(mGuard);
// Update statistics
if (mLastAddTime == 0.0)
mLastAddTime = now_ms();
else
{
float t = now_ms();
mStat.mPacketInterval.process(t - mLastAddTime);
mLastAddTime = t;
}
mStat.mSsrc = static_cast<uint16_t>(packet->GetSSRC());
// Update jitter
@@ -121,15 +139,15 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
// New sequence number
unsigned newSeqno = packet->GetExtendedSequenceNumber();
for (Packet& p: mPacketList)
for (std::shared_ptr<Packet>& p: mPacketList)
{
unsigned seqno = p.rtp()->GetExtendedSequenceNumber();
unsigned seqno = p->rtp()->GetExtendedSequenceNumber();
if (seqno == newSeqno)
{
mStat.mDuplicatedRtp++;
ICELogMedia(<< "Discovered duplicated packet, skipping");
return false;
return std::shared_ptr<Packet>();
}
if (seqno > maxno)
@@ -143,8 +161,11 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
if (newSeqno > minno || (available < mHigh))
{
Packet p(packet, timelength, rate);
// Insert into queue
auto p = std::make_shared<Packet>(packet, timelength, rate);
mPacketList.push_back(p);
// Sort again
std::sort(mPacketList.begin(), mPacketList.end(), SequenceSort);
// Limit by max timelength
@@ -152,21 +173,18 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
if (available > mHigh)
ICELogMedia(<< "Available " << available << "ms with limit " << mHigh << "ms");
/*while (available > mHigh && mPacketList.size())
{
ICELogDebug( << "Dropping RTP packet from jitter buffer");
available -= mPacketList.front().timelength();
mPacketList.erase(mPacketList.begin());
}*/
return p;
}
else
{
ICELogMedia(<< "Too old packet, skipping");
mStat.mOldRtp++;
return false;
return std::shared_ptr<Packet>();
}
return true;
return std::shared_ptr<Packet>();
}
RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
@@ -182,7 +200,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
while (total > mHigh && mPacketList.size())
{
ICELogMedia( << "Dropping RTP packets from jitter buffer");
total -= mPacketList.front().timelength();
total -= mPacketList.front()->timelength();
// Save it as last packet however - to not confuse loss packet counter
mFetchedPacket = mPacketList.front();
@@ -198,7 +216,11 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
result = FetchResult::NoPacket;
else
{
if (mFetchedPacket.rtp())
bool is_fetched_packet = mFetchedPacket.get() != nullptr;
if (is_fetched_packet)
is_fetched_packet &= mFetchedPacket->rtp().get() != nullptr;
if (is_fetched_packet)
{
if (mPacketList.empty())
{
@@ -208,10 +230,10 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
else
{
// Current sequence number ?
unsigned seqno = mPacketList.front().rtp()->GetExtendedSequenceNumber();
unsigned seqno = mPacketList.front()->rtp()->GetExtendedSequenceNumber();
// Gap between new packet and previous on
int gap = seqno - mFetchedPacket.rtp()->GetSequenceNumber() - 1;
int gap = seqno - mFetchedPacket->rtp()->GetSequenceNumber() - 1;
gap = std::min(gap, 127);
if (gap > 0 && mPacketList.empty())
{
@@ -228,29 +250,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
}
result = FetchResult::RegularPacket;
Packet& p = mPacketList.front();
rl.push_back(p.rtp());
// Maybe it is time to replay packet right now ? For case of AMR SID packets
/*if (mFetchedPacket.rtp() && gap == 0 && mFetchedPacket.timelength() >= 10 && p.timelength() >= 10)
{
int timestampDelta;
// Timestamp difference
if (p.rtp()->GetTimestamp() > mFetchedPacket.rtp()->GetTimestamp())
timestampDelta = TimeHelper::getDelta(p.rtp()->GetTimestamp(), mFetchedPacket.rtp()->GetTimestamp());
else
timestampDelta = TimeHelper::getDelta(mFetchedPacket.rtp()->GetTimestamp(), p.rtp()->GetTimestamp());
// Timestamp units per packet
int nrOfPackets = timestampDelta / (p.timelength() * (p.rate() / 1000));
// Add more copies of SID (most probably) packets
for (int i = 0; i < nrOfPackets - 1; i++)
{
//assert(false);
rl.push_back(p.rtp());
}
}*/
rl.push_back(mPacketList.front());
// Save last returned normal packet
mFetchedPacket = mPacketList.front();
@@ -269,7 +269,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
result = FetchResult::RegularPacket;
// Put it to output list
rl.push_back(mPacketList.front().rtp());
rl.push_back(mPacketList.front());
// Remember returned packet
mFetchedPacket = mPacketList.front();
@@ -295,7 +295,7 @@ int RtpBuffer::findTimelength()
{
int available = 0;
for (unsigned i = 0; i < mPacketList.size(); i++)
available += mPacketList[i].timelength();
available += mPacketList[i]->timelength();
return available;
}
@@ -321,7 +321,7 @@ Receiver::~Receiver()
//-------------- AudioReceiver ----------------
AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics &stat)
:Receiver(stat), mBuffer(stat), mFrameCount(0), mFailedCount(0), mCodecSettings(settings),
:Receiver(stat), mBuffer(stat), mCodecSettings(settings),
mCodecList(settings)
{
// Init resamplers
@@ -354,9 +354,45 @@ AudioReceiver::~AudioReceiver()
mDecodedDump.reset();
}
size_t decode_packet(Codec& codec, RTPPacket& p, void* output_buffer, size_t output_capacity)
{
// How much data was produced
size_t result = 0;
// Handle here regular RTP packets
// Check if payload length is ok
int tail = codec.rtpLength() ? p.GetPayloadLength() % codec.rtpLength() : 0;
if (!tail)
{
// Find number of frames
int frame_count = codec.rtpLength() ? p.GetPayloadLength() / codec.rtpLength() : 1;
int frame_length = codec.rtpLength() ? codec.rtpLength() : (int)p.GetPayloadLength();
// Save last packet time length
// mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
// Decode
for (int i=0; i < frame_count; i++)
{
auto decoded_length = codec.decode(p.GetPayloadData() + i * codec.rtpLength(),
frame_length,
output_buffer,
output_capacity);
result += decoded_length;
}
}
else
ICELogMedia(<< "RTP packet with tail.");
return result;
}
bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec)
{
// ICELogInfo(<< "Adding packet No " << p->GetSequenceNumber());
// Increase codec counter
mStat.mCodecCount[p->GetPayloadType()]++;
@@ -410,7 +446,22 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** co
}
// Queue packet to buffer
return mBuffer.add(p, time_length, codecIter->second->samplerate());
auto packet = mBuffer.add(p, time_length, codecIter->second->samplerate()).get();
if (packet)
{
// Check if early decoding configured
if (mEarlyDecode && *codec)
{
// Move data to packet buffer
size_t available = decode_packet(**codec, *p, mDecodedFrame, sizeof mDecodedFrame);
packet->pcm().resize(available / 2);
memcpy(packet->pcm().data(), mDecodedFrame, available / 2);
}
return true;
}
else
return false;
}
void AudioReceiver::processDecoded(Audio::DataWindow& output, int options)
@@ -435,7 +486,7 @@ void AudioReceiver::processDecoded(Audio::DataWindow& output, int options)
bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
{
bool result = false;
bool result = false, /*had_cng = false, */had_decode = false;
// Get next packet from buffer
RtpBuffer::ResultList rl;
@@ -443,7 +494,7 @@ bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
switch (fr)
{
case RtpBuffer::FetchResult::Gap:
ICELogInfo(<< "Gap detected.");
ICELogDebug(<< "Gap detected.");
mDecodedLength = mResampledLength = 0;
if (mCngPacket && mCodec)
@@ -482,14 +533,14 @@ bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
case RtpBuffer::FetchResult::RegularPacket:
mFailedCount = 0;
for (std::shared_ptr<RTPPacket>& p: rl)
for (std::shared_ptr<RtpBuffer::Packet>& p: rl)
{
assert(p);
// Check if previously CNG packet was detected. Emit CNG audio here if needed.
if (options & DecodeOptions_FillCngGap && mCngPacket && mCodec)
{
// Fill CNG audio is server mode is present
int units = p->GetTimestamp() - mCngPacket->GetTimestamp();
int units = p->rtp()->GetTimestamp() - mCngPacket->GetTimestamp();
int milliseconds = units / (mCodec->samplerate() / 1000);
if (milliseconds > mLastPacketTimeLength)
{
@@ -522,69 +573,83 @@ bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
}
}
// Find codec
mCodec = mCodecMap[p->GetPayloadType()];
if (mCodec)
if (mEarlyDecode)
{
if (rate)
*rate = mCodec->samplerate();
// ToDo - copy the decoded data to output buffer
// Check if it is CNG packet
if ((p->GetPayloadType() == 0 || p->GetPayloadType() == 8) && p->GetPayloadLength() >= 1 && p->GetPayloadLength() <= 6)
}
else
{
// Find codec by payload type
int ptype = p->rtp()->GetPayloadType();
mCodec = mCodecMap[ptype];
if (mCodec)
{
if (options & DecodeOptions_SkipDecode)
mDecodedLength = 0;
else
if (rate)
*rate = mCodec->samplerate();
// Check if it is CNG packet
if ((ptype == 0 || ptype == 8) && p->rtp()->GetPayloadLength() >= 1 && p->rtp()->GetPayloadLength() <= 6)
{
mCngPacket = p;
mCngDecoder.decode3389(p->GetPayloadData(), p->GetPayloadLength());
// Emit CNG mLastPacketLength milliseconds
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength,
(short*)mDecodedFrame, true);
if (mDecodedLength)
processDecoded(output, options);
}
result = true;
}
else
{
// Reset CNG packet
mCngPacket.reset();
// Handle here regular RTP packets
// Check if payload length is ok
int tail = mCodec->rtpLength() ? p->GetPayloadLength() % mCodec->rtpLength() : 0;
if (!tail)
{
// Find number of frames
mFrameCount = mCodec->rtpLength() ? p->GetPayloadLength() / mCodec->rtpLength() : 1;
int frameLength = mCodec->rtpLength() ? mCodec->rtpLength() : (int)p->GetPayloadLength();
// Save last packet time length
mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
// Decode
for (int i=0; i<mFrameCount && !mCodecSettings.mSkipDecode; i++)
if (options & DecodeOptions_SkipDecode)
mDecodedLength = 0;
else
{
if (options & DecodeOptions_SkipDecode)
mDecodedLength = 0;
else
{
// Decode frame by frame
mDecodedLength = mCodec->decode(p->GetPayloadData() + i*mCodec->rtpLength(),
frameLength, mDecodedFrame, sizeof mDecodedFrame);
if (mDecodedLength)
processDecoded(output, options);
}
mCngPacket = p->rtp();
mCngDecoder.decode3389(p->rtp()->GetPayloadData(), p->rtp()->GetPayloadLength());
// Emit CNG mLastPacketLength milliseconds
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength,
(short*)mDecodedFrame, true);
if (mDecodedLength)
processDecoded(output, options);
}
result = mFrameCount > 0;
// Check for bitrate counter
processStatisticsWithAmrCodec(mCodec.get());
result = true;
}
else
ICELogMedia(<< "RTP packet with tail.");
{
// Reset CNG packet
mCngPacket.reset();
// Handle here regular RTP packets
// Check if payload length is ok
int tail = mCodec->rtpLength() ? p->rtp()->GetPayloadLength() % mCodec->rtpLength() : 0;
if (!tail)
{
// Find number of frames
mFrameCount = mCodec->rtpLength() ? p->rtp()->GetPayloadLength() / mCodec->rtpLength() : 1;
int frameLength = mCodec->rtpLength() ? mCodec->rtpLength() : (int)p->rtp()->GetPayloadLength();
// Save last packet time length
mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
// Decode
for (int i=0; i<mFrameCount && !mCodecSettings.mSkipDecode; i++)
{
if (options & DecodeOptions_SkipDecode)
mDecodedLength = 0;
else
{
// Trigger the statistics
had_decode = true;
// Decode frame by frame
mDecodedLength = mCodec->decode(p->rtp()->GetPayloadData() + i * mCodec->rtpLength(),
frameLength, mDecodedFrame, sizeof mDecodedFrame);
// mDecodedLength = 3840; // Opus 20 ms stereo
if (mDecodedLength)
processDecoded(output, options);
}
}
result = mFrameCount > 0;
// Check for bitrate counter
processStatisticsWithAmrCodec(mCodec.get());
}
else
ICELogMedia(<< "RTP packet with tail.");
}
}
}
}
@@ -594,6 +659,18 @@ bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
assert(0);
}
if (had_decode)
{
// mStat.mDecodeRequested++;
if (mLastDecodeTime == 0.0)
mLastDecodeTime = now_ms();
else
{
float t = now_ms();
mStat.mDecodingInterval.process(t - mLastDecodeTime);
mLastDecodeTime = t;
}
}
return result;
}
@@ -670,12 +747,12 @@ void AudioReceiver::updatePvqa(const void *data, int size)
mPvqaBuffer->addZero(size);
Audio::Format fmt;
int frames = (int)fmt.timeFromSize(mPvqaBuffer->filled()) / (PVQA_INTERVAL * 1000);
int frames = static_cast<int>(fmt.timeFromSize(mPvqaBuffer->filled())) / (PVQA_INTERVAL * 1000);
if (frames > 0)
{
int time4pvqa = (int)(frames * PVQA_INTERVAL * 1000);
int size4pvqa = (int)fmt.sizeFromTime(time4pvqa);
ICELogInfo(<< "PVQA buffer has " << time4pvqa << " milliseconds of audio.");
ICELogDebug(<< "PVQA buffer has " << time4pvqa << " milliseconds of audio.");
mPVQA->update(mPvqaBuffer->data(), size4pvqa);
mPvqaBuffer->erase(size4pvqa);
}

46
src/engine/media/MT_AudioReceiver.h
View File

@@ -47,12 +47,17 @@ namespace MT
public:
Packet(const std::shared_ptr<RTPPacket>& packet, int timelen, int rate);
std::shared_ptr<RTPPacket> rtp() const;
int timelength() const;
int rate() const;
const std::vector<short>& pcm() const;
std::vector<short>& pcm();
protected:
std::shared_ptr<RTPPacket> mRtp;
int mTimelength = 0, mRate = 0;
std::vector<short> mPcm;
};
RtpBuffer(Statistics& stat);
@@ -60,35 +65,48 @@ namespace MT
unsigned ssrc();
void setSsrc(unsigned ssrc);
void setHigh(int milliseconds);
int high();
void setLow(int milliseconds);
int low();
void setPrebuffer(int milliseconds);
int prebuffer();
int getNumberOfReturnedPackets() const;
int getNumberOfAddPackets() const;
int findTimelength();
int getCount() const;
// Returns false if packet was not add - maybe too old or too new or duplicate
bool add(std::shared_ptr<RTPPacket> packet, int timelength, int rate);
typedef std::vector<std::shared_ptr<RTPPacket>> ResultList;
// Returns false if packet was not add - maybe too old or too new or duplicate
std::shared_ptr<Packet> add(std::shared_ptr<RTPPacket> packet, int timelength, int rate);
typedef std::vector<std::shared_ptr<Packet>> ResultList;
typedef std::shared_ptr<ResultList> PResultList;
FetchResult fetch(ResultList& rl);
protected:
unsigned mSsrc;
int mHigh, mLow, mPrebuffer;
int mReturnedCounter, mAddCounter;
unsigned mSsrc = 0;
int mHigh = RTP_BUFFER_HIGH,
mLow = RTP_BUFFER_LOW,
mPrebuffer = RTP_BUFFER_PREBUFFER;
int mReturnedCounter = 0,
mAddCounter = 0;
mutable Mutex mGuard;
typedef std::vector<Packet> PacketList;
typedef std::vector<std::shared_ptr<Packet>> PacketList;
PacketList mPacketList;
Statistics& mStat;
bool mFirstPacketWillGo;
bool mFirstPacketWillGo = true;
jrtplib::RTPSourceStats mRtpStats;
Packet mFetchedPacket;
std::shared_ptr<Packet> mFetchedPacket;
// To calculate average interval between packet add. It is close to jitter but more useful in debugging.
float mLastAddTime = 0.0;
};
class Receiver
@@ -146,6 +164,9 @@ namespace MT
std::shared_ptr<jrtplib::RTPPacket> mCngPacket;
CngDecoder mCngDecoder;
// Decode RTP early, do not wait for speaker callback
bool mEarlyDecode = false;
// Buffer to hold decoded data
char mDecodedFrame[65536];
int mDecodedLength = 0;
@@ -161,12 +182,17 @@ namespace MT
// Last packet time length
int mLastPacketTimeLength = 0;
int mFailedCount;
int mFailedCount = 0;
Audio::Resampler mResampler8, mResampler16,
mResampler32, mResampler48;
Audio::PWavFileWriter mDecodedDump;
float mLastDecodeTime = 0.0; // Time last call happened to codec->decode()
float mIntervalSum = 0.0;
int mIntervalCount = 0;
// Zero rate will make audio mono but resampling will be skipped
void makeMonoAndResample(int rate, int channels);

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

@@ -97,7 +97,7 @@ AudioStream::~AudioStream()
if (mFinalStatistics)
*mFinalStatistics = mStat;
ICELogInfo(<< mStat.toShortString());
ICELogInfo(<< mStat.toString());
}
void AudioStream::setDestination(const RtpPair<InternetAddress>& dest)

4
src/engine/media/MT_Box.cpp
View File

@@ -34,7 +34,7 @@ Terminal::~Terminal()
mAudioPair.reset();
}
PStream Terminal::createStream(int type, VariantMap& config)
PStream Terminal::createStream(int type, VariantMap& /*config*/)
{
PStream result;
switch (type)
@@ -52,7 +52,7 @@ PStream Terminal::createStream(int type, VariantMap& config)
return result;
}
void Terminal::freeStream(PStream stream)
void Terminal::freeStream(const PStream& stream)
{
if (AudioStream* audio = dynamic_cast<AudioStream*>(stream.get()))
{

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

@@ -26,10 +26,11 @@ namespace MT
CodecList& codeclist();
PStream createStream(int type, VariantMap& config);
void freeStream(PStream s);
void freeStream(const PStream& s);
Audio::PDevicePair audio();
void setAudio(const Audio::PDevicePair& audio);
protected:
StreamList mAudioList;
std::mutex mAudioListMutex;

5
src/engine/media/MT_Codec.h
View File

@@ -63,8 +63,13 @@ public:
virtual int channels() { return 1; }
// Returns size of encoded data (RTP) in bytes
virtual int encode(const void* input, int inputBytes, void* output, int outputCapacity) = 0;
// Returns size of decoded data (PCM signed short) in bytes
virtual int decode(const void* input, int inputBytes, void* output, int outputCapacity) = 0;
// Returns size of produced data (PCM signed short) in bytes
virtual int plc(int lostFrames, void* output, int outputCapacity) = 0;
// Returns size of codec in memory

37
src/engine/media/MT_CodecList.cpp
View File

@@ -25,6 +25,42 @@ using namespace MT;
using strx = StringHelper;
// ---------------- EvsSpec ---------------
std::string CodecList::Settings::toString() const
{
std::ostringstream oss;
oss << "wrap IuUP: " << mWrapIuUP << std::endl
<< "skip decode: " << mSkipDecode << std::endl;
for (int ptype: mAmrWbPayloadType)
oss << "AMR WB ptype: " << ptype << std::endl;
for (int ptype: mAmrWbOctetPayloadType)
oss << "AMR WB octet-aligned ptype: " << ptype << std::endl;
for (int ptype: mAmrNbPayloadType)
oss << "AMR NB ptype: " << ptype << std::endl;
for (int ptype: mAmrNbOctetPayloadType)
oss << "AMR NB octet-aligned ptype:" << ptype << std::endl;
oss << "ISAC 16Khz ptype: " << mIsac16KPayloadType << std::endl
<< "ISAC 32Khz ptype: " << mIsac32KPayloadType << std::endl
<< "iLBC 20ms ptype: " << mIlbc20PayloadType << std::endl
<< "iLBC 30ms ptype: " << mIlbc30PayloadType << std::endl
<< "GSM FR ptype: " << mGsmFrPayloadType << ", GSM FR plength: " << mGsmFrPayloadLength << std::endl
<< "GSM HR ptype: " << mGsmHrPayloadType << std::endl
<< "GSM EFR ptype: " << mGsmEfrPayloadType << std::endl;
for (auto& spec: mEvsSpec)
{
oss << "EVS ptype: " << spec.mPayloadType << ", bw: " << spec.mBandwidth << ", enc: " << (spec.mEncodingType == EvsSpec::Encoding_MIME ? "mime" : "g192") << std::endl;
}
for (auto& spec: mOpusSpec)
{
oss << "OPUS ptype: " << spec.mPayloadType << ", rate: " << spec.mRate << ", channels: " << spec.mChannels << std::endl;
}
return oss.str();
}
bool CodecList::Settings::EvsSpec::isValid() const
{
return mPayloadType >= 96 && mPayloadType <= 127;
@@ -125,7 +161,6 @@ CodecList::CodecList(const Settings& settings)
mFactoryList.push_back(new GsmEfrCodec::GsmEfrFactory(mSettings.mWrapIuUP, mSettings.mGsmEfrPayloadType));
#endif
#endif
mFactoryList.push_back(new IsacCodec::IsacFactory16K(mSettings.mIsac16KPayloadType));
mFactoryList.push_back(new IlbcCodec::IlbcFactory(mSettings.mIlbc20PayloadType, mSettings.mIlbc30PayloadType));
mFactoryList.push_back(new G711Codec::AlawFactory());

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

@@ -82,6 +82,9 @@ public:
};
std::vector<OpusSpec> mOpusSpec;
// Textual representation - used in logging
std::string toString() const;
static Settings DefaultSettings;
};

1
src/engine/media/MT_SingleAudioStream.cpp
View File

@@ -19,6 +19,7 @@ SingleAudioStream::SingleAudioStream(const CodecList::Settings& codecSettings, S
SingleAudioStream::~SingleAudioStream()
{
}
void SingleAudioStream::process(const std::shared_ptr<jrtplib::RTPPacket>& packet)

10
src/engine/media/MT_Statistics.cpp
View File

@@ -191,6 +191,9 @@ Statistics& Statistics::operator += (const Statistics& src)
mJitter = src.mJitter;
mRttDelay = src.mRttDelay;
mDecodingInterval = src.mDecodingInterval;
mDecodeRequested = src.mDecodeRequested;
if (!src.mCodecName.empty())
mCodecName = src.mCodecName;
@@ -239,13 +242,16 @@ Statistics& Statistics::operator -= (const Statistics& src)
}
std::string Statistics::toShortString() const
std::string Statistics::toString() const
{
std::ostringstream oss;
oss << "Received: " << mReceivedRtp
<< ", lost: " << mPacketLoss
<< ", dropped: " << mPacketDropped
<< ", sent: " << mSentRtp;
<< ", sent: " << mSentRtp
<< ", decoding interval: " << mDecodingInterval.average()
<< ", decode requested: " << mDecodeRequested.average()
<< ", packet interval: " << mPacketInterval.average();
return oss.str();
}

82
src/engine/media/MT_Statistics.h
View File

@@ -6,6 +6,8 @@
#include "audio/Audio_DataWindow.h"
#include "helper/HL_Optional.hpp"
#include "helper/HL_Statistics.h"
#include "jrtplib/src/rtptimeutilities.h"
#include "jrtplib/src/rtppacket.h"
@@ -13,78 +15,6 @@ using std::experimental::optional;
namespace MT
{
template<typename T>
struct Average
{
int mCount = 0;
T mSum = 0;
T average() const
{
if (!mCount)
return 0;
return mSum / mCount;
}
T value() const
{
return average();
}
void process(T value)
{
mCount++;
mSum += value;
}
};
template<typename T, int minimum = 100000, int maximum = 0, int default_value = 0>
struct TestResult
{
T mMin = minimum;
T mMax = maximum;
Average<T> mAverage;
T mCurrent = default_value;
void process(T value)
{
if (mMin > value)
mMin = value;
if (mMax < value)
mMax = value;
mCurrent = value;
mAverage.process(value);
}
bool is_initialized() const
{
return mAverage.mCount > 0;
}
T current() const
{
if (is_initialized())
return mCurrent;
else
return 0;
}
T value() const
{
return current();
}
TestResult<T>& operator = (T value)
{
process(value);
return *this;
}
operator T()
{
return mCurrent;
}
};
template<typename T>
struct StreamStats
@@ -130,9 +60,13 @@ public:
mDuplicatedRtp, // Number of received duplicated rtp packets
mOldRtp, // Number of late rtp packets
mPacketLoss, // Number of lost packets
mPacketDropped, // Number of dropped packets (due to time unsync when playing)
mPacketDropped, // Number of dropped packets (due to time unsync when playing)б
mIllegalRtp; // Number of rtp packets with bad payload type
TestResult<float> mDecodingInterval, // Average interval on call to packet decode
mDecodeRequested, // Average amount of requested audio frames to play
mPacketInterval; // Average interval between packet adding to jitter buffer
int mLoss[128]; // Every item is number of loss of corresping length
size_t mAudioTime; // Decoded/found time in milliseconds
uint16_t mSsrc; // Last known SSRC ID in a RTP stream
@@ -169,7 +103,7 @@ public:
std::string mPvqaReport;
#endif
std::string toShortString() const;
std::string toString() const;
};
} // end of namespace MT

10
src/engine/media/MT_Stream.cpp
View File

@@ -21,7 +21,7 @@ Stream::Stream()
Stream::~Stream()
{
ICELogInfo(<< mStat.toString());
}
void Stream::setDestination(const RtpPair<InternetAddress>& dest)
@@ -77,13 +77,13 @@ StreamList::~StreamList()
clear();
}
void StreamList::add(PStream s)
void StreamList::add(const PStream& s)
{
Lock l(mMutex);
mStreamVector.push_back(s);
}
void StreamList::remove(PStream s)
void StreamList::remove(const PStream& s)
{
Lock l(mMutex);
@@ -98,7 +98,7 @@ void StreamList::clear()
mStreamVector.clear();
}
bool StreamList::has(PStream s)
bool StreamList::has(const PStream& s)
{
Lock l(mMutex);
return std::find(mStreamVector.begin(), mStreamVector.end(), s) != mStreamVector.end();
@@ -127,4 +127,4 @@ void StreamList::copyTo(StreamList* sl)
Mutex& StreamList::getMutex()
{
return mMutex;
}
}

6
src/engine/media/MT_Stream.h
View File

@@ -88,10 +88,10 @@ namespace MT
StreamList();
~StreamList();
void add(PStream s);
void remove(PStream s);
void add(const PStream& s);
void remove(const PStream& s);
void clear();
bool has(PStream s);
bool has(const PStream& s);
int size();
PStream streamAt(int index);