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- ongoing work to fix issue with unstable audio

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This commit is contained in:
Dmytro Bogovych
2021-08-06 18:31:42 +03:00
parent ed39725641
commit 74b5aa69cb
18 changed files with 220 additions and 119 deletions
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37
src/engine/media/MT_AudioReceiver.cpp
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@@ -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>
@@ -108,7 +109,16 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
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
@@ -357,6 +367,7 @@ AudioReceiver::~AudioReceiver()
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()]++;
@@ -435,7 +446,7 @@ 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 +454,7 @@ 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)
@@ -571,9 +582,13 @@ bool AudioReceiver::getAudio(Audio::DataWindow& output, int options, int* rate)
mDecodedLength = 0;
else
{
// Trigger the statistics
had_decode = true;
// Decode frame by frame
mDecodedLength = mCodec->decode(p->GetPayloadData() + i*mCodec->rtpLength(),
mDecodedLength = mCodec->decode(p->GetPayloadData() + i * mCodec->rtpLength(),
frameLength, mDecodedFrame, sizeof mDecodedFrame);
// mDecodedLength = 3840; // Opus 20 ms stereo
if (mDecodedLength)
processDecoded(output, options);
}
@@ -594,6 +609,18 @@ 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 +697,12 @@ 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);
}