| 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
|
|
|