Chromium Code Reviews| Index: webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
| index 5fc68e65222897e871d06f826ec604b95d387d9a..89878b7e457dba46d920d5c5c527f9ec43bf8446 100644 |
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h |
| @@ -31,6 +31,11 @@ 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. Note that the default value of this variable may need to be updated |
|
ossu
2017/09/14 12:16:14
"Note that the default value of this variable may
ivoc
2017/09/14 13:03:39
I guess this is a bit arbitrary and probably doesn
|
| + // if the default frame length is increased. |
| + bool last_fl_change_increase = false; |
| }; |
| } // namespace webrtc |