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- clang format + naming improved

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This commit is contained in:
Dmytro Bogovych
2026-09-03 10:03:42 +03:00
parent 249e614def
commit e166bae1ea
204 changed files with 20621 additions and 21019 deletions
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src/engine/media/MT_AudioReceiver.cpp
+142 -121
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@@ -16,8 +16,9 @@
#include <cmath>
#include <iostream>
#if !defined(TARGET_ANDROID) && !defined(TARGET_OPENWRT) && !defined(TARGET_WIN) && !defined(TARGET_RPI) && defined(USE_AMR_CODEC)
# include "MT_AmrCodec.h"
#if !defined(TARGET_ANDROID) && !defined(TARGET_OPENWRT) && !defined(TARGET_WIN) && !defined(TARGET_RPI) && \
defined(USE_AMR_CODEC)
#include "MT_AmrCodec.h"
#endif
#include <algorithm>
@@ -27,9 +28,11 @@
using namespace MT;
// ----------------- RtpBuffer::Packet --------------
RtpBuffer::Packet::Packet(const std::shared_ptr<RTPPacket>& packet, std::chrono::milliseconds timelength, int samplerate)
:mRtp(packet), mTimelength(timelength), mSamplerate(samplerate)
{}
RtpBuffer::Packet::Packet(const std::shared_ptr<RTPPacket>& packet, std::chrono::milliseconds timelength,
int samplerate)
: mRtp(packet), mTimelength(timelength), mSamplerate(samplerate)
{
}
const std::shared_ptr<RTPPacket>& RtpBuffer::Packet::rtp() const
{
@@ -57,15 +60,13 @@ std::vector<short>& RtpBuffer::Packet::pcm()
}
// ------------ RtpBuffer ----------------
RtpBuffer::RtpBuffer(Statistics& stat)
:mStat(stat)
{
}
RtpBuffer::RtpBuffer(Statistics& stat) : mStat(stat) {}
RtpBuffer::~RtpBuffer()
{
if (mAddCounter)
ICELogDebug(<< "Number of add packets: " << mAddCounter << ", number of retrieved packets " << mReturnedCounter);
ICELogDebug(<< "Number of add packets: " << mAddCounter << ", number of retrieved packets "
<< mReturnedCounter);
}
void RtpBuffer::setHigh(std::chrono::milliseconds t)
@@ -109,7 +110,8 @@ bool SequenceSort(const std::shared_ptr<RtpBuffer::Packet>& p1, const std::share
return p1->rtp()->GetExtendedSequenceNumber() < p2->rtp()->GetExtendedSequenceNumber();
}
std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(const std::shared_ptr<jrtplib::RTPPacket>& packet, std::chrono::milliseconds timelength, int rate)
std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(const std::shared_ptr<jrtplib::RTPPacket>& packet,
std::chrono::milliseconds timelength, int rate)
{
if (!packet)
return std::shared_ptr<Packet>();
@@ -137,7 +139,7 @@ std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(const std::shared_ptr<jrtplib:
// New sequence number
unsigned newSeqno = packet->GetExtendedSequenceNumber();
for (auto& p: mPacketList)
for (auto& p : mPacketList)
{
unsigned seqno = p->rtp()->GetExtendedSequenceNumber();
@@ -196,16 +198,15 @@ void RtpBuffer::trimToHighWater(size_t maxPackets)
// high-water mark (mHigh, when set) or, if maxPackets != 0, the packet-count
// cap. Always keep at least one packet so loss/gap accounting has a reference.
while (mPacketList.size() > 1 &&
((0ms != mHigh && total > mHigh) ||
(maxPackets != 0 && mPacketList.size() > maxPackets)))
((0ms != mHigh && total > mHigh) || (maxPackets != 0 && mPacketList.size() > maxPackets)))
{
ICELogMedia( << "Dropping RTP packets from jitter buffer");
ICELogMedia(<< "Dropping RTP packets from jitter buffer");
total -= mPacketList.front()->timelength();
// Before advancing mLastSeqno over the dropped packet, record a loss event for any
// sequence-number gap on the wire between the previous packet we saw and this one.
// Without this, drops silently mask real packet loss that happened between them.
auto droppingPacket = mPacketList.front();
auto droppingPacket = mPacketList.front();
uint32_t droppingSeq = droppingPacket->rtp()->GetExtendedSequenceNumber();
if (mLastSeqno)
{
@@ -241,7 +242,7 @@ void RtpBuffer::trimToHighWater(size_t maxPackets)
RtpBuffer::FetchResult RtpBuffer::fetch()
{
Lock l(mGuard);
Lock l(mGuard);
FetchResult result;
@@ -268,11 +269,11 @@ RtpBuffer::FetchResult RtpBuffer::fetch()
else
{
// Current sequence number ?
auto& packet = *mPacketList.front();
auto& packet = *mPacketList.front();
uint32_t seqno = packet.rtp()->GetExtendedSequenceNumber();
// Gap between new packet and previous on
int gap = (int64_t)seqno - (int64_t)*mLastSeqno - 1;
int gap = (int64_t)seqno - (int64_t)*mLastSeqno - 1;
if (gap > 0)
{
// std::cout << "Increase the packet loss for SSRC " << std::hex << mSsrc << std::endl;
@@ -290,8 +291,9 @@ RtpBuffer::FetchResult RtpBuffer::fetch()
.mTimestampEnd = gapEnd});
}
// ToDo: here we should decide smth - 2-packet gap shoud report Status::Gap two times at least; but current implementation gives only one.
// It is not big problem - as gap is detected when we have smth to return usually
// ToDo: here we should decide smth - 2-packet gap shoud report Status::Gap two times at least; but
// current implementation gives only one. It is not big problem - as gap is detected when we have
// smth to return usually
mLastSeqno = seqno;
mLastReceiveTime = packet.rtp()->GetReceiveTime();
result = {FetchResult::Status::Gap};
@@ -343,7 +345,7 @@ RtpBuffer::FetchResult RtpBuffer::fetch()
std::chrono::milliseconds RtpBuffer::findTimelength()
{
std::chrono::milliseconds r = 0ms;
for (const auto& p: mPacketList)
for (const auto& p : mPacketList)
r += p->timelength();
return r;
}
@@ -356,9 +358,9 @@ std::shared_ptr<RtpBuffer::Packet> RtpBuffer::peekFront() const
std::chrono::milliseconds RtpBuffer::bufferedTime() const
{
Lock l(mGuard);
Lock l(mGuard);
std::chrono::milliseconds r = 0ms;
for (const auto& p: mPacketList)
for (const auto& p : mPacketList)
r += p->timelength();
return r;
}
@@ -374,16 +376,14 @@ int RtpBuffer::getNumberOfAddPackets() const
}
//-------------- Receiver ---------------
Receiver::Receiver(Statistics& stat)
:mStat(stat)
{}
Receiver::Receiver(Statistics& stat) : mStat(stat) {}
Receiver::~Receiver()
{}
Receiver::~Receiver() {}
//-------------- AudioReceiver ----------------
AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics &stat)
:Receiver(stat), mRtpBuffer(stat), mDtmfBuffer(stat), mCodecSettings(settings), mCodecList(settings), mDtmfReceiver(stat)
AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics& stat)
: Receiver(stat), mRtpBuffer(stat), mDtmfBuffer(stat), mCodecSettings(settings), mCodecList(settings),
mDtmfReceiver(stat)
{
// Init codecs
mCodecList.setSettings(settings);
@@ -449,7 +449,8 @@ void AudioReceiver::setCodecSettings(const CodecList::Settings& codecSettings)
const bool lazy = mCodecSettings.mLazyCodecMap;
mCodecSettings = codecSettings;
mCodecSettings.mLazyCodecMap = lazy;
mCodecList.setSettings(mCodecSettings); // This builds factory list with proper payload types according to payload types in settings
mCodecList.setSettings(
mCodecSettings); // This builds factory list with proper payload types according to payload types in settings
// Rebuild codec map from factory list
mCodecList.fillCodecMap(mCodecMap);
@@ -485,10 +486,7 @@ Codec* AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p)
Codec* codec = nullptr;
// Estimate time length
int time_length = 0,
samplerate = 8000,
payloadLength = p->GetPayloadLength(),
ptype = p->GetPayloadType();
int time_length = 0, samplerate = 8000, payloadLength = p->GetPayloadLength(), ptype = p->GetPayloadType();
// ICELogMedia(<< "Adding packet No " << p->GetSequenceNumber());
@@ -522,8 +520,7 @@ Codec* AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p)
if (!codec)
time_length = 10;
else
if (!codec->rtpLength())
else if (!codec->rtpLength())
time_length = codec->frameTime();
else
time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
@@ -543,14 +540,14 @@ Codec* AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p)
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 - smth strange with found codec...
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
mRtpBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
return nullptr;
}
// Check if packet is too short from time length side - smth strange with found codec...
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
mRtpBuffer.add(p, std::chrono::milliseconds(time_length), samplerate);
return nullptr;
}
// Queue packet to buffer
mRtpBuffer.add(p, std::chrono::milliseconds(time_length), samplerate).get();
@@ -618,7 +615,8 @@ void AudioReceiver::produceCNG(std::chrono::milliseconds length, Audio::DataWind
if (options.mSkipDecode)
mDecodedLength = 0;
else
mDecodedLength = cng().produce(mCodec->samplerate(), tail, reinterpret_cast<short*>(mDecodedFrame.data()), false);
mDecodedLength =
cng().produce(mCodec->samplerate(), tail, reinterpret_cast<short*>(mDecodedFrame.data()), false);
if (mDecodedLength)
processDecoded(output, options);
@@ -636,13 +634,13 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output
{
// Synthesize comfort noise. It will be done on AUDIO_SAMPLERATE rate directly to mResampledFrame buffer.
// Do not forget to send this noise to analysis
mDecodedLength = cng().produce(mCodec->samplerate(), mLastPacketTimeLength, reinterpret_cast<short*>(mDecodedFrame.data()), false);
mDecodedLength = cng().produce(mCodec->samplerate(), mLastPacketTimeLength,
reinterpret_cast<short*>(mDecodedFrame.data()), false);
}
else
decodePacketTo(output, options, mCngPacket);
}
else
if (mCodec && mFrameCount && !mCodecSettings.mSkipDecode)
else if (mCodec && mFrameCount && !mCodecSettings.mSkipDecode)
{
// Conceal the gap into mDecodedFrame/mDecodedLength.
if (options.mSkipDecode)
@@ -662,8 +660,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output
auto codecIter = mCodecMap.find(next->rtp()->GetPayloadType());
if (codecIter != mCodecMap.end() && codecIter->second == mCodec)
{
mDecodedLength = mCodec->fec({next->rtp()->GetPayloadData(), (size_t)next->rtp()->GetPayloadLength()},
concealBuffer);
mDecodedLength = mCodec->fec(
{next->rtp()->GetPayloadData(), (size_t)next->rtp()->GetPayloadLength()}, concealBuffer);
}
}
@@ -675,8 +673,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output
if (!mDecodedLength)
{
// PLC unsupported or failed - substitute one frame of silence.
size_t frameBytes = (size_t)mCodec->frameTime() * mCodec->samplerate() / 1000
* sizeof(short) * std::max(1, mCodec->channels());
size_t frameBytes = (size_t)mCodec->frameTime() * mCodec->samplerate() / 1000 * sizeof(short) *
std::max(1, mCodec->channels());
frameBytes = std::min(frameBytes, concealBuffer.size_bytes());
mDecodedLength = frameBytes;
memset(mDecodedFrame.data(), 0, mDecodedLength);
@@ -703,48 +701,51 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output
if (mDecodedLength)
{
processDecoded(output, options);
return {.mStatus = DecodeResult::Status::Ok, .mSamplerate = mCodec->samplerate(), .mChannels = mCodec->channels()};
return {
.mStatus = DecodeResult::Status::Ok, .mSamplerate = mCodec->samplerate(), .mChannels = mCodec->channels()};
}
else
return {.mStatus = DecodeResult::Status::Skip};
}
AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& output, DecodeOptions options, const std::shared_ptr<RtpBuffer::Packet>& packet)
AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& output, DecodeOptions options,
const std::shared_ptr<RtpBuffer::Packet>& packet)
{
if (!packet || !packet->rtp())
return {DecodeResult::Status::Skip};
DecodeResult result = {.mStatus = DecodeResult::Status::Skip};
auto& rtp = *packet->rtp(); // Syntax sugar
auto& rtp = *packet->rtp(); // Syntax sugar
mFailedCount = 0;
// Check if we need to emit silence - it may happen in the case if next packet has RTP timestamp much beyond the previous one; maybe DTX was active.
// Check if we need to emit silence - it may happen in the case if next packet has RTP timestamp much beyond the
// previous one; maybe DTX was active.
if (mLastPacketTimestamp && mLastPacketTimeLength && mCodec)
{
int units = rtp.GetTimestamp() - *mLastPacketTimestamp;
int milliseconds = units / (mCodec->samplerate() / 1000);
if (milliseconds > mLastPacketTimeLength)
{
auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
ICELogDebug(<< "Emit " << silenceLength << " silence while requested " << options.mElapsed);
silenceLength = std::min(silenceLength, options.mElapsed);
if (mCngPacket && options.mFillGapByCNG)
produceCNG(silenceLength, output, options);
else
produceSilence(silenceLength, output, options);
}
}
int units = rtp.GetTimestamp() - *mLastPacketTimestamp;
int milliseconds = units / (mCodec->samplerate() / 1000);
if (milliseconds > mLastPacketTimeLength)
{
auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
ICELogDebug(<< "Emit " << silenceLength << " silence while requested " << options.mElapsed);
silenceLength = std::min(silenceLength, options.mElapsed);
if (mCngPacket && options.mFillGapByCNG)
produceCNG(silenceLength, output, options);
else
produceSilence(silenceLength, output, options);
}
}
mLastPacketTimestamp = rtp.GetTimestamp();
// Find codec by payload type
int ptype = rtp.GetPayloadType();
int ptype = rtp.GetPayloadType();
// Look into mCodecMap if exists
auto codecIter = mCodecMap.find(ptype);
if (codecIter == mCodecMap.end())
return {};
return {};
if (!codecIter->second)
codecIter->second = mCodecList.createCodecByPayloadType(ptype);
@@ -756,7 +757,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
result.mSamplerate = mCodec->samplerate();
// Check if it is CNG packet
if (((ptype == 0 || ptype == 8) && rtp.GetPayloadLength() >= 1 && rtp.GetPayloadLength() <= 6) || rtp.GetPayloadType() == 13)
if (((ptype == 0 || ptype == 8) && rtp.GetPayloadLength() >= 1 && rtp.GetPayloadLength() <= 6) ||
rtp.GetPayloadType() == 13)
{
if (options.mSkipDecode)
mDecodedLength = 0;
@@ -767,7 +769,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
cng().decode3389(rtp.GetPayloadData(), rtp.GetPayloadLength());
// Emit CNG mLastPacketLength milliseconds
mDecodedLength = cng().produce(mCodec->samplerate(), mLastPacketTimeLength, (short*)mDecodedFrame.data(), true);
mDecodedLength =
cng().produce(mCodec->samplerate(), mLastPacketTimeLength, (short*)mDecodedFrame.data(), true);
if (mDecodedLength)
processDecoded(output, options);
}
@@ -783,7 +786,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
size_t payload_length = rtp.GetPayloadLength();
size_t rtp_frame_length = mCodec->rtpLength();
int tail = rtp_frame_length ? payload_length % rtp_frame_length : 0;
int tail = rtp_frame_length ? payload_length % rtp_frame_length : 0;
if (!tail)
{
@@ -795,15 +798,17 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
mLastPacketTimeLength = mFrameCount * mCodec->frameTime();
// Decode
for (int i=0; i<mFrameCount && !mCodecSettings.mSkipDecode; i++)
for (int i = 0; i < mFrameCount && !mCodecSettings.mSkipDecode; i++)
{
if (options.mSkipDecode)
mDecodedLength = 0;
else
{
// Decode frame by frame
auto codecInput = std::span{rtp.GetPayloadData() + i * mCodec->rtpLength(), (size_t)frameLength};
auto codecOutput = std::span{(uint8_t*)mDecodedFrame.data(), mDecodedFrame.size() * sizeof(int16_t)};
auto codecInput =
std::span{rtp.GetPayloadData() + i * mCodec->rtpLength(), (size_t)frameLength};
auto codecOutput =
std::span{(uint8_t*)mDecodedFrame.data(), mDecodedFrame.size() * sizeof(int16_t)};
auto r = mCodec->decode(codecInput, codecOutput);
mDecodedLength = r.mDecoded;
if (mDecodedLength > 0)
@@ -832,8 +837,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
AudioReceiver::DecodeResult AudioReceiver::decodeEmptyTo(Audio::DataWindow& output, DecodeOptions options)
{
// There are two cases
// First is we have no ready time estimated how much audio should be emitted i.e. audio is decoded right after the next packet arrives.
// In this case we just skip the analysis - we should not be called in this situation
// First is we have no ready time estimated how much audio should be emitted i.e. audio is decoded right after the
// next packet arrives. In this case we just skip the analysis - we should not be called in this situation
if (options.mElapsed == 0ms || !mCodec)
return {.mStatus = DecodeResult::Status::Skip};
@@ -851,7 +856,8 @@ AudioReceiver::DecodeResult AudioReceiver::decodeEmptyTo(Audio::DataWindow& outp
// the CNG decoder writes straight into its buffer.
size_t bytesPerMs = (size_t)fmt.rate() / 1000 * sizeof(short) * fmt.channels();
size_t room = output.capacity() - output.filled();
int ms = bytesPerMs ? (int)std::min<int64_t>(options.mElapsed.count(), (int64_t)(room / bytesPerMs)) : 0;
int ms =
bytesPerMs ? (int)std::min<int64_t>(options.mElapsed.count(), (int64_t)(room / bytesPerMs)) : 0;
if (ms <= 0)
return {.mStatus = DecodeResult::Status::Skip};
auto produced = cng().produce(fmt.rate(), ms, (short*)(output.mutableData() + output.filled()), false);
@@ -918,11 +924,8 @@ void MT::AudioReceiver::logReceiveHeartbeat()
size_t dropDelta = mStat.mPacketDropped - mStatDropLast;
ICELogInfo(<< "[rx-heartbeat] buffered=" << mRtpBuffer.bufferedTime().count() << "ms"
<< " packets=" << mRtpBuffer.getCount()
<< " underruns/5s=" << mUnderrunCount
<< " gaps/5s=" << mGapCount
<< " loss/5s=" << lossDelta
<< " dropped/5s=" << dropDelta);
<< " packets=" << mRtpBuffer.getCount() << " underruns/5s=" << mUnderrunCount << " gaps/5s=" << mGapCount
<< " loss/5s=" << lossDelta << " dropped/5s=" << dropDelta);
mStatLogLast = t;
mUnderrunCount = 0;
@@ -938,7 +941,8 @@ void MT::AudioReceiver::updateDecodingTimeStatistics()
else
{
auto t = std::chrono::steady_clock::now();
mStat.mDecodingInterval.process(std::chrono::duration_cast<std::chrono::milliseconds>(t - *mDecodeTimestamp).count());
mStat.mDecodingInterval.process(
std::chrono::duration_cast<std::chrono::milliseconds>(t - *mDecodeTimestamp).count());
mDecodeTimestamp = t;
}
}
@@ -946,7 +950,7 @@ void MT::AudioReceiver::updateDecodingTimeStatistics()
AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output, DecodeOptions options)
{
// ICELogDebug(<< "getAudioTo() for " << options.mElapsed);
assert (options.mElapsed != 0ms);
assert(options.mElapsed != 0ms);
// First decode on this receiver: allocate the scratch buffers. Network-MOS-only
// streams never reach this point, so they never pay for them.
@@ -961,7 +965,7 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
processDtmf();
// How much time length audio we produced here
auto produced = 0ms;
auto produced = 0ms;
Audio::Format fmt;
// Have we anything from the previous decode attempts ?
@@ -993,9 +997,17 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
// Decode to mAvailable buffer
switch (fr.mStatus)
{
case RtpBuffer::FetchResult::Status::Gap: mGapCount++; result = decodeGapTo(mAvailable, options.decreaseElapsedBy(produced)); break;
case RtpBuffer::FetchResult::Status::NoPacket: result = decodeEmptyTo(mAvailable, options.decreaseElapsedBy(produced)); break;
case RtpBuffer::FetchResult::Status::RegularPacket: result = decodePacketTo(mAvailable, options.decreaseElapsedBy(produced), fr.mPacket); updateDecodeIntervalStatistics(); break;
case RtpBuffer::FetchResult::Status::Gap:
mGapCount++;
result = decodeGapTo(mAvailable, options.decreaseElapsedBy(produced));
break;
case RtpBuffer::FetchResult::Status::NoPacket:
result = decodeEmptyTo(mAvailable, options.decreaseElapsedBy(produced));
break;
case RtpBuffer::FetchResult::Status::RegularPacket:
result = decodePacketTo(mAvailable, options.decreaseElapsedBy(produced), fr.mPacket);
updateDecodeIntervalStatistics();
break;
default:
assert(0);
}
@@ -1005,8 +1017,8 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
break; // No sense to continue - we have no information at all
fmt = options.mResampleToMainRate ? Audio::Format(AUDIO_SAMPLERATE, 1) : mCodec->getAudioFormat();
result.mSamplerate = fmt.rate();
result.mChannels = fmt.channels();
result.mSamplerate = fmt.rate();
result.mChannels = fmt.channels();
// How much milliseconds we have in audio buffer ?
auto bufferAvailable = mAvailable.getTimeLength(fmt);
@@ -1026,7 +1038,8 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
}
mProducedAudio += produced;
// ICELogDebug(<< "Requested " << options.mElapsed << ", produced " << produced << ", remains " << mAvailable.getTimeLength(fmt) << ", packets " << getRtpBuffer().getCount());
// ICELogDebug(<< "Requested " << options.mElapsed << ", produced " << produced << ", remains " <<
// mAvailable.getTimeLength(fmt) << ", packets " << getRtpBuffer().getCount());
logReceiveHeartbeat();
return result;
@@ -1059,21 +1072,31 @@ void AudioReceiver::makeMonoAndResample(int rate, int channels)
if (channels != AUDIO_CHANNELS)
{
if (channels == 1)
mConvertedLength = Audio::ChannelConverter::monoToStereo(mDecodedFrame.data(), mDecodedLength, mConvertedFrame.data(), mDecodedLength * 2);
mConvertedLength = Audio::ChannelConverter::monoToStereo(mDecodedFrame.data(), mDecodedLength,
mConvertedFrame.data(), mDecodedLength * 2);
else
mDecodedLength = Audio::ChannelConverter::stereoToMono(mDecodedFrame.data(), mDecodedLength, mDecodedFrame.data(), mDecodedLength / 2);
mDecodedLength = Audio::ChannelConverter::stereoToMono(mDecodedFrame.data(), mDecodedLength,
mDecodedFrame.data(), mDecodedLength / 2);
}
void* frames = mConvertedLength ? (void*)mConvertedFrame.data() : (void*)mDecodedFrame.data();
unsigned length = mConvertedLength ? mConvertedLength : mDecodedLength;
void* frames = mConvertedLength ? (void*)mConvertedFrame.data() : (void*)mDecodedFrame.data();
unsigned length = mConvertedLength ? mConvertedLength : mDecodedLength;
Audio::Resampler* r = nullptr;
switch (rate)
{
case 8000: r = &mResampler8; break;
case 16000: r = &mResampler16; break;
case 32000: r = &mResampler32; break;
case 48000: r = &mResampler48; break;
case 8000:
r = &mResampler8;
break;
case 16000:
r = &mResampler16;
break;
case 32000:
r = &mResampler32;
break;
case 48000:
r = &mResampler48;
break;
default:
memcpy(mResampledFrame.data(), frames, length);
mResampledLength = length;
@@ -1081,7 +1104,8 @@ void AudioReceiver::makeMonoAndResample(int rate, int channels)
}
size_t processedInput = 0;
mResampledLength = r->processBuffer(frames, length, processedInput, mResampledFrame.data(), r->getDestLength(length));
mResampledLength =
r->processBuffer(frames, length, processedInput, mResampledFrame.data(), r->getDestLength(length));
// processedInput result value is ignored - it is always equal to length as internal sample rate is 8/16/32/48K
}
@@ -1093,7 +1117,8 @@ Codec* AudioReceiver::findCodec(int payloadType)
void AudioReceiver::updateAmrCodecStats(Codec* c)
{
#if !defined(TARGET_ANDROID) && !defined(TARGET_OPENWRT) && !defined(TARGET_WIN) && !defined(TARGET_RPI) && defined(USE_AMR_CODEC)
#if !defined(TARGET_ANDROID) && !defined(TARGET_OPENWRT) && !defined(TARGET_WIN) && !defined(TARGET_RPI) && \
defined(USE_AMR_CODEC)
AmrNbCodec* nb = dynamic_cast<AmrNbCodec*>(c);
AmrWbCodec* wb = dynamic_cast<AmrWbCodec*>(c);
@@ -1102,8 +1127,7 @@ void AudioReceiver::updateAmrCodecStats(Codec* c)
mStat.mBitrateSwitchCounter = nb->getSwitchCounter();
mStat.mCng = nb->getCngCounter();
}
else
if (wb != nullptr)
else if (wb != nullptr)
{
mStat.mBitrateSwitchCounter = wb->getSwitchCounter();
mStat.mCng = wb->getCngCounter();
@@ -1114,7 +1138,8 @@ void AudioReceiver::updateAmrCodecStats(Codec* c)
int AudioReceiver::getSize() const
{
int result = 0;
result += sizeof(*this) + mResampler8.getSize() + mResampler16.getSize() + mResampler32.getSize() + mResampler48.getSize();
result += sizeof(*this) + mResampler8.getSize() + mResampler16.getSize() + mResampler32.getSize() +
mResampler48.getSize();
if (mCodec)
; // ToDo: need the way to calculate size of codec instances
@@ -1133,18 +1158,17 @@ AudioReceiver::MediaInfo AudioReceiver::infoFor(jrtplib::RTPPacket& p)
if (codec->rtpLength() != 0)
{
int frameCount = static_cast<int>(p.GetPayloadLength() / codec->rtpLength());
if (p.GetPayloadType() == 9/*G729A silence*/ && p.GetPayloadLength() % codec->rtpLength())
if (p.GetPayloadType() == 9 /*G729A silence*/ && p.GetPayloadLength() % codec->rtpLength())
frameCount++;
packetTime = std::chrono::milliseconds(frameCount * codec->frameTime());
}
else
if (typeid(*codec) == typeid(OpusCodec))
else if (typeid(*codec) == typeid(OpusCodec))
{
OpusCodec* oc = dynamic_cast<OpusCodec*>(codec);
assert(oc);
size_t samplesCount = oc->getNumberOfSamples({p.GetPayloadData(), p.GetPayloadLength()});
int sampleratePerMs = codec->samplerate() / 1000;
int sampleratePerMs = codec->samplerate() / 1000;
packetTime = std::chrono::milliseconds(samplesCount / sampleratePerMs);
}
else
@@ -1161,18 +1185,15 @@ void AudioReceiver::updateDecodeIntervalStatistics()
if (mLastDecodeTimestamp)
{
mIntervalBetweenDecode += std::chrono::duration_cast<std::chrono::microseconds>(now - *mLastDecodeTimestamp);
mDecodeCount ++;
mDecodeCount++;
}
mLastDecodeTimestamp = now;
}
// ----------------------- DtmfReceiver -------------------
DtmfReceiver::DtmfReceiver(Statistics& stat)
:Receiver(stat)
{}
DtmfReceiver::DtmfReceiver(Statistics& stat) : Receiver(stat) {}
DtmfReceiver::~DtmfReceiver()
{}
DtmfReceiver::~DtmfReceiver() {}
void DtmfReceiver::add(const std::shared_ptr<RTPPacket>& p)
{
@@ -1192,8 +1213,8 @@ void DtmfReceiver::add(const std::shared_ptr<RTPPacket>& p)
mCallback(ev.mTone);
// Queue statistics item
mStat.mDtmf2833Timeline.emplace_back(Dtmf2833Event{.mTone = ev.mTone,
.mTimestamp = RtpHelper::toMicroseconds(p->GetReceiveTime())});
mStat.mDtmf2833Timeline.emplace_back(
Dtmf2833Event{.mTone = ev.mTone, .mTimestamp = RtpHelper::toMicroseconds(p->GetReceiveTime())});
}
mEvent = ev.mTone;