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- G.722: decode as 16 kHz audio, keep the 8 kHz RTP clock separate

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G722Codec::info() reported an 8 kHz sample rate for its 16 kHz output (640
bytes of PCM per 20 ms frame, read as 40 ms at 8 kHz). AudioReceiver therefore
fetched packets at half the rate they arrived, and the high-water trim
discarded about 44% of every G.722 stream: 109 of 250 packets per 5 s, with the
buffer at 360 ms instead of 100 ms.

The 8 kHz figure is right for one thing: RFC 3551 keeps G.722's RTP clock at
8000. Codec::Info already had mTimestampUnit for that. G.722 now reports
mSamplerate 16000 and mTimestampUnit 1/8000, and Codec::rtpClockRate() gives
the clock rate. The uses of samplerate() that mean RTP time now use it:
- the jitter statistics in AudioReceiver::add;
- the timestamp-gap-to-milliseconds conversion in AudioReceiver;
- AudioStream's transmit timestamp unit and increment (numerically unchanged
  for every codec, including G.722).
PCM uses stay on samplerate(), so AudioStream now also feeds the G.722
encoder 16 kHz audio instead of 8 kHz. The factory still advertises G722/8000
in SDP, as it must; only its backwards comment is fixed.

AudioReceiver::infoFor() also applied a "G729A silence" rule to payload type
9, which is G.722; G.729 is 18. It now applies to PT 18. Other fixed-frame
payloads are timed proportionally, as AudioReceiver::add already does, so a
10 ms G.722 packet (half a 160-byte frame) counts as 10 ms, not 20.

Measured in vq-core with 16 replayed calls (NISQA speech; G.722 from ffmpeg
and pjmedia, G.711 controls of the same speech; 3 runs each, all identical):
- G.722 packet drops: 44% -> 0; buffer 100 ms like G.711; no underruns;
- Sevana MOS of every G.722 call now equals its G.711 control, normal and loud
  (with the previous code three calls read 4.45 against 4.25);
- DeadAir-01 0.68-0.89 -> 0.01-0.27 (G.711: 0.00-0.14); SilentCall within 0.01
  of G.711 for the ffmpeg-encoded calls (was 0.15-0.17 higher);
- AmpClipping on loud calls 0.004/0.001, as G.711; the 0.02-0.03 seen after
  the codec swap alone came from the dropped packets;
- G.711 results, packet counts, loss, jitter and audio durations unchanged.
The transmit path (AudioStream) builds but was not exercised; vq-core only
receives.

Needs the vq_net change that computes jitter with rtpClockRate().

Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
This commit is contained in:
Dmytro Bogovych
2026-09-21 18:03:58 +02:00
parent c8d9bad845
commit 03c58c4572
4 changed files with 25 additions and 13 deletions
Download Patch File Download Diff File Expand all files Collapse all files
src/engine/media/MT_AudioCodec.cpp
+7 -4
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@@ -1132,13 +1132,15 @@ G722Codec::~G722Codec()
Codec::Info G722Codec::info()
{
// ToDo: double check the G722 calls - remember RFC has bug about samplerate
// G.722 decodes to 16 kHz (640 bytes of PCM per 20 ms frame), but RFC 3551 keeps its RTP
// clock at 8 kHz for historical reasons: the timestamp unit is 1/8000, not 1/samplerate.
return {.mName = G722_MIME_NAME,
.mSamplerate = 8000,
.mSamplerate = 16000,
.mChannels = 1,
.mPcmLength = 640,
.mFrameTime = 20,
.mRtpLength = 160};
.mRtpLength = 160,
.mTimestampUnit = 1.0f / 8000};
}
Codec::EncodeResult G722Codec::encode(std::span<const uint8_t> input, std::span<uint8_t> output)
@@ -1186,7 +1188,8 @@ const char* G722Codec::G722Factory::name()
int G722Codec::G722Factory::samplerate()
{
// Although G722 uses 16000 as rate for timestamping RTP frames - in fact it is 8KHz codec. So return 8KHz here.
// The rate SDP advertises (rtpmap G722/8000) is the RTP clock, 8 kHz by RFC 3551, although the
// audio is 16 kHz; see G722Codec::info().
return 8000;
}
src/engine/media/MT_AudioReceiver.cpp
+11 -6
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@@ -526,7 +526,7 @@ Codec* AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p)
time_length = lround(double(payloadLength) / codec->rtpLength() * codec->frameTime());
if (codec)
samplerate = codec->samplerate();
samplerate = codec->rtpClockRate(); // RTP timestamp units, not PCM rate (G.722)
}
// Process jitter anyway - can we decode payload or not
@@ -724,7 +724,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacketTo(Audio::DataWindow& out
if (mLastPacketTimestamp && mLastPacketTimeLength && mCodec)
{
int units = rtp.GetTimestamp() - *mLastPacketTimestamp;
int milliseconds = units / (mCodec->samplerate() / 1000);
int milliseconds = units / (mCodec->rtpClockRate() / 1000);
if (milliseconds > mLastPacketTimeLength)
{
auto silenceLength = std::chrono::milliseconds(milliseconds - mLastPacketTimeLength);
@@ -1157,12 +1157,17 @@ 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())
frameCount++;
// A G.729 (PT 18) SID frame is shorter than a voice frame but stands for one; other
// codecs scale with the payload (G.722 at 10 ms ptime carries half a 160-byte frame).
if (p.GetPayloadType() == 18 && p.GetPayloadLength() % codec->rtpLength())
{
int frameCount = static_cast<int>(p.GetPayloadLength() / codec->rtpLength()) + 1;
packetTime = std::chrono::milliseconds(frameCount * codec->frameTime());
}
else
packetTime = std::chrono::milliseconds(
lround(double(p.GetPayloadLength()) / codec->rtpLength() * codec->frameTime()));
}
else if (typeid(*codec) == typeid(OpusCodec))
{
OpusCodec* oc = dynamic_cast<OpusCodec*>(codec);
src/engine/media/MT_AudioStream.cpp
+2 -2
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@@ -114,7 +114,7 @@ void AudioStream::setTransmittingCodec(Codec::Factory& factory, int payloadType)
mTransmittingCodec = factory.create();
mTransmittingPayloadType = payloadType;
if (mRtpSession.IsActive())
mRtpSession.SetTimestampUnit(1.0 / mTransmittingCodec->samplerate());
mRtpSession.SetTimestampUnit(mTransmittingCodec->timestampUnit());
}
PCodec AudioStream::transmittingCodec()
@@ -241,7 +241,7 @@ void AudioStream::addData(const void* buffer, int bytes)
ICELogMedia(<< "Sending RTP packet pt = " << mTransmittingPayloadType
<< ", plength = " << (int)mEncodedAudio.size() << " to ");
mRtpSession.SendPacketEx(mEncodedAudio.data(), mEncodedAudio.size(), mTransmittingPayloadType, false,
packetTime * codec->samplerate() / 1000, 0, nullptr, 0);
packetTime * codec->rtpClockRate() / 1000, 0, nullptr, 0);
mEncodedAudio.clear();
encodedTime = 0;
}
src/engine/media/MT_Codec.h
+4
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@@ -66,6 +66,10 @@ public:
int frameTime() { return info().mFrameTime; }
std::string name() { return info().mName; }
float timestampUnit() { return info().mTimestampUnit == 0.0f ? 1.0f / info().mSamplerate : info().mTimestampUnit; }
// RTP clock rate in Hz, the unit of RTP timestamps. Equals samplerate() except for codecs
// whose RTP clock differs from the decoded audio: G.722 decodes to 16 kHz but its RTP clock
// is 8 kHz (RFC 3551). Use it for timestamp arithmetic and jitter, samplerate() for PCM.
int rtpClockRate() { return static_cast<int>(1.0f / timestampUnit() + 0.5f); }
Audio::Format getAudioFormat() { return Audio::Format(this->info().mSamplerate, this->info().mChannels); }