Chromium Code Reviews| Index: webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h |
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h |
| index 82baa60b8fa18fe355d162c136ba7f815f2565a4..d19710254d23bc3d997546d3ccaf74d4911d2684 100644 |
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h |
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h |
| @@ -34,9 +34,14 @@ class FrameLengthController final : public Controller { |
| ~Config(); |
| std::vector<int> encoder_frame_lengths_ms; |
| int initial_frame_length_ms; |
| + // Uplink packet loss fraction below which frame length can increase. |
|
minyue-webrtc
2016/09/28 12:47:26
added together with the comments in config.proto.
|
| float fl_increasing_packet_loss_fraction; |
| + // Uplink packet loss fraction below which frame length should decrease. |
| float fl_decreasing_packet_loss_fraction; |
| + // Uplink bandwidth below which frame length can switch from 20ms to 60ms. |
| int fl_20ms_to_60ms_bandwidth_bps; |
| + // Uplink bandwidth above which frame length should switch from 60ms to |
| + // 20ms. |
| int fl_60ms_to_20ms_bandwidth_bps; |
| }; |