Chromium Code Reviews| Index: webrtc/call.h |
| diff --git a/webrtc/call.h b/webrtc/call.h |
| index 313c5e58c16f49446c8d6fbcdaa92950179628a8..a195aee5d3a3219579dd89d724e6c10363170b4c 100644 |
| --- a/webrtc/call.h |
| +++ b/webrtc/call.h |
| @@ -17,6 +17,7 @@ |
| #include "webrtc/audio_receive_stream.h" |
| #include "webrtc/audio_send_stream.h" |
| #include "webrtc/audio_state.h" |
| +#include "webrtc/base/optional.h" |
| #include "webrtc/base/socket.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| @@ -77,7 +78,7 @@ class Call { |
| struct BitrateConfig { |
| int min_bitrate_bps = 0; |
| int start_bitrate_bps = kDefaultStartBitrateBps; |
| - int max_bitrate_bps = -1; |
| + rtc::Optional<int> max_bitrate_bps; |
| } bitrate_config; |
|
pthatcher1
2016/03/17 21:51:28
Should we just use kMaxBandwidth here and not have
skvlad
2016/03/18 00:49:17
It appears that the kMaxBandwidth value = 2Mbps is
|
| // AudioState which is possibly shared between multiple calls. |