- fix packet time calculation for Opus codec

src/engine/media/MT_AudioCodec.cpp

@@ -549,6 +549,12 @@ size_t OpusCodec::plc(int lostPackets, std::span<uint8_t> output)
     return ((uint8_t*)data_output - output.data());
 }
 
+size_t OpusCodec::getNumberOfSamples(std::span<const uint8_t> payload)
+{
+    int r = opus_packet_get_nb_samples(payload.data(), payload.size(), mSamplerate);
+    return r >= 0 ? r : 0;
+}
+
 // -------------- ILBC -------------------
 #define ILBC_CODEC_NAME "ILBC"
 

src/engine/media/MT_AudioCodec.h

@@ -64,8 +64,11 @@ class OpusCodec: public Codec
 protected:
     OpusEncoder        *mEncoderCtx = nullptr;
     OpusDecoder        *mDecoderCtx = nullptr;
-    int mPTime = 0, mSamplerate = 0, mChannels = 0;
+    int mPTime = 0,
+        mSamplerate = 0,
+        mChannels = 0;
     int mDecoderChannels = 0;
+
 public:
     struct Params
     {
@@ -110,6 +113,8 @@ public:
     EncodeResult    encode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
     DecodeResult    decode(std::span<const uint8_t> input, std::span<uint8_t> output) override;
     size_t          plc(int lostFrames, std::span<uint8_t> output) override;
+
+    size_t          getNumberOfSamples(std::span<const uint8_t> payload);
 };
 
 

src/engine/media/MT_AudioReceiver.cpp

@@ -936,18 +936,31 @@ AudioReceiver::MediaInfo AudioReceiver::infoFor(jrtplib::RTPPacket& p)
     if (!codec)
         return {};
 
-    int frame_count = 0;
+    std::chrono::milliseconds packetTime = 0ms;
+
     if (codec->rtpLength() != 0)
     {
-        frame_count = static_cast<int>(p.GetPayloadLength() / codec->rtpLength());
+        int frameCount = static_cast<int>(p.GetPayloadLength() / codec->rtpLength());
         if (p.GetPayloadType() == 9/*G729A silence*/ && p.GetPayloadLength() % codec->rtpLength())
-            frame_count++;
+            frameCount++;
+
+        packetTime = std::chrono::milliseconds(frameCount * codec->frameTime());
     }
     else
-        frame_count = 1;
+    if (typeid(*codec) == typeid(OpusCodec))
+    {
+        OpusCodec* oc = dynamic_cast<OpusCodec*>(codec.get());
+        assert(oc);
+        size_t samplesCount = oc->getNumberOfSamples({p.GetPayloadData(), p.GetPayloadLength()});
+        int sampleratePerMs = codec->samplerate() / 1000;
+        packetTime = std::chrono::milliseconds(samplesCount / sampleratePerMs);
+    }
+    else
+    {
+        packetTime = std::chrono::milliseconds(codec->frameTime());
+    }
 
-
+    return {packetTime, codec->samplerate()};
-    return {std::chrono::milliseconds(frame_count * codec->frameTime()), codec->samplerate()};
 }
 
 // int AudioReceiver::timelengthFor(jrtplib::RTPPacket& p)