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- performance optimizations

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This commit is contained in:
Dmytro Bogovych
2026-06-17 14:18:31 +03:00
parent 40f3b34b16
commit 2cb2a93d59
7 changed files with 177 additions and 32 deletions
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src/engine/media/MT_AudioReceiver.cpp
+53 -21
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@@ -184,16 +184,18 @@ std::shared_ptr<RtpBuffer::Packet> RtpBuffer::add(const std::shared_ptr<jrtplib:
return std::shared_ptr<Packet>();
}
RtpBuffer::FetchResult RtpBuffer::fetch()
void RtpBuffer::trimToHighWater(size_t maxPackets)
{
Lock l(mGuard);
FetchResult result;
// See if there is enough information in buffer
auto total = findTimelength();
while (total > mHigh && mPacketList.size() > 1 && 0ms != mHigh)
// Drop the oldest packet while either bound is exceeded: the time-based
// 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)))
{
ICELogMedia( << "Dropping RTP packets from jitter buffer");
total -= mPacketList.front()->timelength();
@@ -233,6 +235,19 @@ RtpBuffer::FetchResult RtpBuffer::fetch()
// Increase number in statistics
mStat.mPacketDropped++;
}
}
RtpBuffer::FetchResult RtpBuffer::fetch()
{
Lock l(mGuard);
FetchResult result;
// Bound the buffer to the high-water mark before fetching.
trimToHighWater();
// See how much audio is buffered now.
auto total = findTimelength();
if (total < mLow || total == 0ms)
{
@@ -494,13 +509,13 @@ void AudioReceiver::processDecoded(Audio::DataWindow& output, DecodeOptions opti
{
// Write to audio dump if requested
if (mDecodedDump && mDecodedLength)
mDecodedDump->write(mDecodedFrame, mDecodedLength);
mDecodedDump->write(mDecodedFrame.data(), mDecodedLength);
// Resample to target rate
makeMonoAndResample(options.mResampleToMainRate ? mCodec->samplerate() : 0, mCodec->channels());
// Send to output
output.add(mResampledFrame, mResampledLength);
output.add(mResampledFrame.data(), mResampledLength);
}
void AudioReceiver::produceSilence(std::chrono::milliseconds length, Audio::DataWindow& output, DecodeOptions options)
@@ -517,13 +532,13 @@ void AudioReceiver::produceSilence(std::chrono::milliseconds length, Audio::Data
size_t tail_size = tail * sizeof(int16_t) * mCodec->samplerate() / 1000 * mCodec->channels();
for (size_t i = 0; i < chunks; i++)
{
memset(mDecodedFrame, 0, chunk_size);
memset(mDecodedFrame.data(), 0, chunk_size);
mDecodedLength = chunk_size;
processDecoded(output, options);
}
if (tail)
{
memset(mDecodedFrame, 0, tail_size);
memset(mDecodedFrame.data(), 0, tail_size);
mDecodedLength = tail_size;
processDecoded(output, options);
}
@@ -537,7 +552,7 @@ void AudioReceiver::produceCNG(std::chrono::milliseconds length, Audio::DataWind
if (options.mSkipDecode)
mDecodedLength = 0;
else
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), 100, mDecodedFrame, false);
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), 100, mDecodedFrame.data(), false);
if (mDecodedLength)
processDecoded(output, options);
@@ -550,7 +565,7 @@ void AudioReceiver::produceCNG(std::chrono::milliseconds length, Audio::DataWind
if (options.mSkipDecode)
mDecodedLength = 0;
else
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), tail, reinterpret_cast<short*>(mDecodedFrame), false);
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), tail, reinterpret_cast<short*>(mDecodedFrame.data()), false);
if (mDecodedLength)
processDecoded(output, options);
@@ -568,7 +583,7 @@ 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 = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, reinterpret_cast<short*>(mDecodedFrame), false);
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, reinterpret_cast<short*>(mDecodedFrame.data()), false);
}
else
decodePacketTo(output, options, mCngPacket);
@@ -581,14 +596,14 @@ AudioReceiver::DecodeResult AudioReceiver::decodeGapTo(Audio::DataWindow& output
mDecodedLength = 0;
else
{
mDecodedLength = mCodec->plc(mFrameCount, {(uint8_t*)mDecodedFrame, sizeof mDecodedFrame});
mDecodedLength = mCodec->plc(mFrameCount, {(uint8_t*)mDecodedFrame.data(), mDecodedFrame.size() * sizeof(int16_t)});
if (!mDecodedLength)
{
// PLC is not support or failed
// So substitute the silence
size_t nr_of_samples = mCodec->frameTime() * mCodec->samplerate() / 1000 * sizeof(short);
mDecodedLength = nr_of_samples * sizeof(short);
memset(mDecodedFrame, 0, mDecodedLength);
memset(mDecodedFrame.data(), 0, mDecodedLength);
}
}
}
@@ -660,7 +675,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
mCngDecoder.decode3389(rtp.GetPayloadData(), rtp.GetPayloadLength());
// Emit CNG mLastPacketLength milliseconds
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, (short*)mDecodedFrame, true);
mDecodedLength = mCngDecoder.produce(mCodec->samplerate(), mLastPacketTimeLength, (short*)mDecodedFrame.data(), true);
if (mDecodedLength)
processDecoded(output, options);
}
@@ -696,7 +711,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
{
// Decode frame by frame
auto codecInput = std::span{rtp.GetPayloadData() + i * mCodec->rtpLength(), (size_t)frameLength};
auto codecOutput = std::span{(uint8_t*)mDecodedFrame, sizeof mDecodedFrame};
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)
@@ -798,6 +813,10 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
// ICELogDebug(<< "getAudioTo() for " << options.mElapsed);
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.
ensureDecodeBuffers();
// Increase counter of requested audio
mRequestedAudio += options.mElapsed;
@@ -876,6 +895,19 @@ AudioReceiver::DecodeResult AudioReceiver::getAudioTo(Audio::DataWindow& output,
return result;
}
void AudioReceiver::ensureDecodeBuffers()
{
// Allocate the decode/convert/resample scratch buffers to full capacity on the
// first decode. mDecodedFrame being empty means none are allocated yet; they
// are always allocated together, so checking one is enough.
if (mDecodedFrame.empty())
{
mDecodedFrame.resize(MT_MAX_DECODEBUFFER);
mConvertedFrame.resize(MT_MAX_DECODEBUFFER * 2);
mResampledFrame.resize(MT_MAX_DECODEBUFFER);
}
}
void AudioReceiver::makeMonoAndResample(int rate, int channels)
{
// Make mono from stereo - engine works with mono only for now
@@ -883,12 +915,12 @@ void AudioReceiver::makeMonoAndResample(int rate, int channels)
if (channels != AUDIO_CHANNELS)
{
if (channels == 1)
mConvertedLength = Audio::ChannelConverter::monoToStereo(mDecodedFrame, mDecodedLength, mConvertedFrame, mDecodedLength * 2);
mConvertedLength = Audio::ChannelConverter::monoToStereo(mDecodedFrame.data(), mDecodedLength, mConvertedFrame.data(), mDecodedLength * 2);
else
mDecodedLength = Audio::ChannelConverter::stereoToMono(mDecodedFrame, mDecodedLength, mDecodedFrame, mDecodedLength / 2);
mDecodedLength = Audio::ChannelConverter::stereoToMono(mDecodedFrame.data(), mDecodedLength, mDecodedFrame.data(), mDecodedLength / 2);
}
void* frames = mConvertedLength ? mConvertedFrame : mDecodedFrame;
void* frames = mConvertedLength ? (void*)mConvertedFrame.data() : (void*)mDecodedFrame.data();
unsigned length = mConvertedLength ? mConvertedLength : mDecodedLength;
Audio::Resampler* r = nullptr;
@@ -899,13 +931,13 @@ void AudioReceiver::makeMonoAndResample(int rate, int channels)
case 32000: r = &mResampler32; break;
case 48000: r = &mResampler48; break;
default:
memcpy(mResampledFrame, frames, length);
memcpy(mResampledFrame.data(), frames, length);
mResampledLength = length;
return;
}
size_t processedInput = 0;
mResampledLength = r->processBuffer(frames, length, processedInput, mResampledFrame, 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
}