Index: modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
index 5b2d11371bfdb0706ff1a6e7a228fc574cd193ac..5af3e09ec827afcd63ee382d3ec5f4fd24c9470a 100644 |
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
@@ -32,6 +32,15 @@ struct AudioEncoderRuntimeConfig { |
// better use of the bandwidth. |num_channels| sets the number of channels |
// to encode. |
rtc::Optional<size_t> num_channels; |
+ |
+ // This is true if the last frame length change was an increase, and otherwise |
+ // false. |
+ // The value of this boolean is used to apply a different offset to the |
+ // per-packet overhead that is reported by the BWE. The exact offset value |
+ // is most important right after a frame length change, because the frame |
+ // length change affects the overhead. In the steady state, the exact value is |
+ // not important because the BWE will compensate. |
+ bool last_fl_change_increase = false; |
}; |
} // namespace webrtc |